[OpenSIPS-Users] acc record outbound response to originator

[Dave Singer]

I'm wanting to record the response sent back to the call originator, not
just the last winning response selected by tm module. There are cases where
opensips translates like a received 503 to a 500 or like if you don't trust
the response you received from a vendor that is not using sip codes
properly.
Also want to log when I reject a call without making any outbound attempts.
Like for loop detected or some other invalid formatting. It seems it only
logs after t_relay has been used.

-- 
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Re: [OpenSIPS-Users] Forward() question

[Dave Singer]

Luis,

forward() does not accept a variable for an argument.
instead use like:
   $du = "sip:" + src_ip + ":5070";
I think it needs URI format without the user portion.
see: http://www.opensips.org/Resources/DocsCoreVar17#toc34

There are a bunch of older functions that do not accept variables as
arguments. Some of them like this have read/write variables that can be used
instead.
It would probably be good to farmiliarize yourself with all the vars on that
page especially the read/write ones.

Dave

On Sun, Oct 2, 2011 at 7:17 PM, Luis Morales
wrote:

>  Greetings,
>
> I'd like to use a variable with the function forward() to pass it the
> destination, but I can't find a way to do it. I've look in the
> documentation, but i don't know if I'm doing something wrong, or it just
> can't be done.
>
> I'd like to something like this:
>
> $var(dest) = src_ip + ":5070";
> forward( $var(dest));
>
>
> Thanks,
> Luis Morales.
>
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Re: [OpenSIPS-Users] need step by step guide lines for media proxy installation

[Dave Singer]

Have you checked if relay and dispatcher are actually running?
Have you run them in the foreground with debug turned on to see why
use_media_proxy returned an error?
I'm betting Brett is right that radius is at least primarily the problem.
I would disable it's use in opensips.cfg and mediaproxy/config.ini to see
where that gets you.
With the instructions that you followed I'm guessing you installed on a
debian based system. I used those instructions along with a couple others to
get it working on CentOS (Red Hat) systems.
I posted my detailed steps
http://opensips.org/pipermail/users/2011-January/016399.html
It might help for working out the kinks of the configs. There are also links
to the other instruction sources I used.

Dave


On Mon, Sep 12, 2011 at 12:47 PM, Akib Sayyed  wrote:

> please i want instruction to install mediaproxy
>
> On Mon, Sep 12, 2011 at 9:58 PM, Adrian Georgescu wrote:
>
>> That web site instructions are completely bogus
>>
>> Adrian
>>
>> On Sep 12, 2011, at 5:39 PM, Akib Sayyed wrote:
>>
>> Guys i want step by step guide for installing and running mediaproxy.
>> i followed
>> http://voiprookie.blogspot.com/2009/04/blog-post.html instructions but i
>> received following error
>>
>>
>> Sep 12 21:06:03 localhost /usr/local/sbin/opensips[15062]: rc_ip_hostname:
>> couldn't look up host by addr: 73FCBDE7
>> Sep 12 21:06:03 localhost /usr/local/sbin/opensips[15062]: rc_send_server:
>> no reply from RADIUS server unknown:1813
>> Sep 12 21:06:03 localhost /usr/local/sbin/opensips[15062]:
>> ERROR:acc:acc_aaa_request: Radius accounting request failed for status:
>> 'Failed' Call-Id:
>> '4006d02bca688a29MDY2NjMyZWU0Njk5ODVlYTkzZWI1ZDliMzA5NDBhYzc.'
>> Sep 12 21:06:08 localhost /usr/local/sbin/opensips[15062]: ACC:
>> transaction answered:
>> timestamp=1315841768;method=BYE;from_tag=21357d7c;to_tag=63261562;call_id=624deb2d230cae3bMDY2NjMyZWU0Njk5ODVlYTkzZWI1ZDliMzA5NDBhYzc.;code=200;reason=OK
>> Sep 12 21:06:16 localhost /usr/local/sbin/opensips[15058]: rc_ip_hostname:
>> couldn't look up host by addr: 73FCBDE7
>> Sep 12 21:06:16 localhost /usr/local/sbin/opensips[15058]: rc_send_server:
>> no reply from RADIUS server unknown:1813
>> Sep 12 21:06:16 localhost /usr/local/sbin/opensips[15058]:
>> ERROR:acc:acc_aaa_request: Radius accounting request failed for status:
>> 'Start' Call-Id:
>> '624deb2d230cae3bMDY2NjMyZWU0Njk5ODVlYTkzZWI1ZDliMzA5NDBhYzc.'
>> Sep 12 21:06:16 localhost /usr/local/sbin/opensips[15063]:
>> ERROR:mediaproxy:use_media_proxy: mediaproxy returned error
>> Sep 12 21:06:17 localhost /usr/local/sbin/opensips[15060]: rc_ip_hostname:
>> couldn't look up host by addr: 73FCBDE7
>> Sep 12 21:06:17 localhost /usr/local/sbin/opensips[15060]: rc_send_server:
>> no reply from RADIUS server unknown:1813
>> Sep 12 21:06:17 localhost /usr/local/sbin/opensips[15060]:
>> ERROR:acc:acc_aaa_request: Radius accounting request failed for status:
>> 'Alive' Call-Id:
>> '624deb2d230cae3bMDY2NjMyZWU0Njk5ODVlYTkzZWI1ZDliMzA5NDBhYzc.'
>>
>>
>> i am using media-dispatcher and media-relay command to run dispatcher and
>> relay resp
>> please let me know
>> thanks in advance
>>
>> --
>> Akib Sayyed
>> Matrix-Shell
>> akibsay...@gmail.com
>> akibsay...@matrixshell.com
>> Mob:- +91-966-514-2243
>>
>>
>> ___
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>>
>>
>>
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>>
>>
>
>
> --
> Akib Sayyed
> Matrix-Shell
> akibsay...@gmail.com
> akibsay...@matrixshell.com
> Mob:- +91-966-514-2243
>
>
>
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>
>


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Re: [OpenSIPS-Users] avp set in branch_route

[Dave Singer]

avps are per transaction. A branch is just part of a transaction.
Until the transaction is terminated the avp stays active. Be aware
that a call's avp is write-able in all branches and assigning a value
to an existing avp pushes the new value on the top of the avp array.
Calls do not go back to request routing after any reply or failure.
Assuming you set to trigger all routing block types as in route below,
the order is route, branch route, reply route, failure route.
route {
  .
  do_routing("1");
  route("my_stuff");
  t_relay;
  exit;
}
on_branch[tb]{
  #each branch runs through here (assuming triggered as above.) after
the exit is hit.
}
on_reply[tr]{
  #all replies received hit here but not communication timeouts.
  #avps written here are not saved unless avpops param is set to
}
on_failure[tf] {
  #all avps (except from on_reply if appropriate avpops param is not
set) and only pseudo vars from the "winning branch" are available
here.
  #from here you can kick off another serial/parallel fork.
  #you can call any custom named route block with route("my_stuff").
  #Be sure that what is in that route block can be used by the
  # primary route mode (REQUEST_ROUTE, ONREPY_ROUTE,
  # FAILURE_ROUTE, etc) it is being called from. Example, you
  # wouldn't call route("my_stuff") (see below) from an ONREPLY_ROUTE
  # block because it contains t_relay which can not be used in that route
  # block mode. Calling from the wrong one will cause config check test
  # to warn of config error.
  if(use_next_gw()) {
route("my_stuff");
  }
}

route[my_stuff] {
  t_on_reply("tr");
  t_on_branch("tb");
  t_on_failure("tf");
  t_relay();
  exit;
}

On Tue, Aug 30, 2011 at 8:54 AM, Jeff Pyle  wrote:
> Hello,
>
> If an avp is set in a branch_route, does that avp still exist if a 
> failure_route catches a negative reply and sends the call back into request 
> routing for a serial fork?  Or, does the avp go away once that branch goes 
> away?
>
>
> - Jeff
>
>
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Re: [OpenSIPS-Users] Preparing new major release 1.7.0 - UPDATE

[Dave Singer]

Speaking of suggestions... (understandable probably won't make it in
1.7, more likely 2.0)
I've been thinking about variables in the script and a couple other things.
1. New variables for direct read/write access to value currently only
available by function (very beneficial for example with cdrs)
a. Dialog variable like $dlg(name)
b. Memcache local like $mcl(name). example usage: if (
$mcl(bad_auth_cnt/$si) == null ) { $mcl(bad_auth_cnt/$si) = 1 } else {
if ( $mcl(bad_auth_cnt/$si) > 10 ) { xlog("WARNING", "Bad password
count exceeded for $si") ; drop; } ; $mcl(bad_auth_cnt/$si) =
$mcl(bad_auth_cnt/$si) + 1 } ; # when var is set it uses default cache
timeout optionally set in core setting section of script ( like listen
option )
c. Memcache regular like $mc(name)
d. Message var that lasts the duration of the mesage like
mvar(name). Fast like var but don't have to worry if it was set by
this message or a previous message.
2. Make all pseudo vars (avp, var, dlg, mc, mcl, mvar) support all
data types like avps:
a. arrays
b. json
c. string
d. int
e. float (new)
f.  hash (new)
3. Make all pseudo vars (avp, var, dlg, mc, mcl, mvar) support all the
functions, operators and assignment capabilities of avps like
avp_db_load, avp_check, and avp_subst. The only difference between avp
and var, dlg etc would be what part of memory they are stored in. This
would be useful for many thing like loading memcache from db.
4. Add function that combines functionality of avp_db_load and
avp_db_query. Eg: var_db_load_query("$mvar(qry)", "$avp(qres)/h"); #
result from db of query from $mvar(qry) is in same format (columns) as
returned for avp_db_load. Results are loaded into $avp(qres) as a
hash. Since it is loaded into a single var results can be loaded
several times into different vars without concern for an avp(name)
already being loaded (so if results say to forward call to PSTN for a
call from PSTN you can reload options for diverting user and not worry
about weather the avp was loaded by the first load.
5. Add an option flag to t_on_failure("1", "m"), m for manual failure
route rather than auto as done with DNS SRV so you can choose weather
to try the next dest according to SRV etc dependent on  response. In
this case it might be beneficial to store the SRV records in a json
(would be able to represent difference between load balance hosts and
fail over host of SRV) or avp.
6. Enhancements for DRouting
a. Cache drouting user/domain group id and attr in mem like rules
and gateways. (Used memcache to speed things up.)
b. Fix do_routing to convert a number in an avp string. (Annoyance
if for example pulled from memcache as string, must be converted to
int before passed to do_routing.)
c. Add support for ability to look up user/domain and
default/domain with user/domain group id and attr being the first
values and default/domain in the result avps if there is a user/domain
match. If you had 50,000 users but only 50 could do international you
could save 49,949 db entries. (I implemented this with a fairly
complex sql query with avp_db_query)
d. Add support to pass "|" separated list of user/domain vars to
use in user/domain look up, first var set without null is used for
look up. Eg: 
do_routing("o/$di|$(ai{uri.user})@$si|$(re{uri.user})@$si|$fU@$si");
could support a short hand like do_routing("SO/darf"); S indicates to
use $si for domain and O indicates the header order to try. ( I
implemented this with checking in order of preference if the various
headers were present setting $avp(s:readSI) and using that in the
avp_db_query used to group id)
e. Add support for aliasing multiple domains to a single alias
domain that so duplication of user/domain settings can be avoided.
Like if you had a 5 proxy servers all setup to send calls from 50,000
users, you would have 250,000. With aliasing you would have only
25,006. ( This I used dalias in the username column of the db and in
the description only put the domain it was aliased to, again adding to
the complexity of the query)
f. Add support for only checking domain. Eg: do_routing("ds"); d
for domain only and s for use $si for domain. ( I implemented this by
using any in the db for the user and added to the complexity of the
query)
g. Add support for re-lookup of source domain according to a
custom header like X-RealSrcIP:192.168.0.1. Might add a column to the
dr_groups table for options and a mod param defining the custom header
name or search the attr column for a key/val pair like XSH=X-RealSrcIP
This is good for your topology hiding servers that would pass the
source IP they got the call from. ( I checked the returned attr avp
for a key/val pair and reran if necessary)
h. Add attr column to dr_gateways. Had to use description column
to store what I would in attr column and assign it to the avp in the
avp_db_query call.

So where are suggestions usually posted? :-)

Thanks again to all who have contributed to

Re: [OpenSIPS-Users] /etc/init.d/opensips restart returns OK when opensips is down...

[Dave Singer]

Toyima,

What I've done to deal with errors on a restart is to copy the config
to a temp, replace a couple things like fifo file and ports, start it
with foreground option, piping output to check for errors. The group
is backgrounded with & and the init script waits for 2 sec. If bk
process is still running and no errors, it is killed and real restart
is proceeded with.

I posted my init script a few months back in the list if you want more details.
Be cautious of the script return value as restart failure may be just
that the config was bad and opensips was not restarted but still
running old config, you may not want to have HA kick to other server.

Dave

On Mon, Jun 6, 2011 at 3:24 PM, Robert Thomas  wrote:
> I have an HA setup with Pacemaker using heartbeat stack.
>
> I created a script that monitors opensips via the pid file. Eventhough the
> start event succeed and then opensips crash due to a module. The next
> monitor round will report an error, and the resource would be migrated to
> the  next node.
>
> I don't think the start stop daemon utility used to bring up opensips would
> succeed, and then opensips would crash starting the modules.
>
> On Mon, Jun 6, 2011 at 10:44 AM, Toyima Dias  wrote:
>>
>> Ok, but it any case, the script should return 1 if opensips failed to
>> initialize; i'm thinking to build a HA infraestructure with opensips
>> (heartbeat, cluster glue...) and this opensips init script behavior
>> seems not to be appropriate; i will have to take a deeper look at
>> the opensips.init script
>>
>> 2011/6/6 Andrew Pogrebennyk 
>>>
>>> On 06.06.2011 14:15, Toyima Dias wrote:

 If opensips has not started (ps -ef | grep opensips doesn't show any
 opensips pid), how is possible that /etc/init.d/opensips returns [OK]
 when you execute /etc/init.d/opensips restart...is this beacuse of
 mysql? is that what you mean...this script should return 1 (not
 succesfull, different than 0) it no pid file has been created, am i
 right?
>>>
>>> Correct, but there can be a case when opensips starts but then some
>>> module's initialization function fails due to mysql connect error or
>>> incompatible table version and the opensips will exit.
>>>
>>> --
>>> Sincerely,
>>> Andrew Pogrebennyk
>>>
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>>
>>
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>
>
>
> --
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>
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Re: [OpenSIPS-Users] multiple subscribers per alias

[Dave Singer]

Erik,

Just typed this up so probably has some mistakes. :-)
This form pulls all alias db results into an avp which are looped
though to do a lookup for each that only a user alias. You could do
the append branch mode for alias_db but then lookup would have to be
in branch route and Im not sure if branches created by lookup would
run through another round of branch route to handle last minute per
branch message settings.

So for what it is worth:

  if( alias_db_find("dbaliases" , "$ru", "$avp(s:ru_alias)", "d") {
$var(append_next) = 0;
$var(i) = 0;
while( $(avp(s:ru_alias)[$var(i)]) != null ) {
   if ( $var(append_next) == 1 )
 append_branch();
   else
 $var(append_next) = 1;
   $ru = $(avp(s:ru_alias)[$var(i)]);
   if ( rui == myself ) {
 if ( ! lookup("locaion") )
   # lookup failed
   $var(append_next) = 0;
   }
   $var(i) = $var(i) + 1;
}
if ( $var(append_next) == 0 ) {
  # user not registered, sent to VM server or reply not found
  sl_send_reply("404", "User not found");
  exit;
}
t_on_branch("branch_changes");
route("1");
  }
  else
  {
  if ( lookup("location") ) {
 

Hope this helps.
Dave

On Wed, May 25, 2011 at 1:17 PM, Erik Dekkers  wrote:
> Hi folks,
>
> I am having trouble finding the best way of solving this situation;
>
> Customer can login via a webinterface to change his/her settings like
> usernames of sip-accounts, passwords, etc.
> The customer would like to change his/her aliasses too. So they can
> configure themself wich sip-account(s) should be used for a specifice
> inbound phonenumber.
>
> In this case it can happen a customer would like to have two sip accounts
> behind one alias (e.g. phonenumber 12345 let ring phones registered with
> account sip1 and sip2).
> The most logical way for me is to configure this within the alias_db module,
> but I can't put in two alias-records with different subscribers.
>
> How should I make this possible without hardcode it in the script?
>
> Kind regards,
>
> Erik Dekkers
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Re: [OpenSIPS-Users] Rewrite Username in URI when using userloc

[Dave Singer]

Spencer,
Looks like you just need to move $rU=$oU; into branch_route[2]
Note that in failure route it will be reset to what it was before
changes done in branch route so you may want to do it in main route
and in branch route.

Dave

On Thu, May 19, 2011 at 4:02 PM, Spencer Thomason
 wrote:
> Hello all,
> I'm using Opensips in between several Freeswitch and Asterisk boxes (each 
> with multiple DIDs) and my ITSP's proxies.  The local IP PBXs register to 
> Opensips and inbound routing is done via the userloc module.  The problem I 
> have is that I need to send in incoming INVITE to the original username, not 
> the registered contact so the PBXs can perform the proper routing.  Currently 
> I set $rU=$Ou in the request route which works perfectly if there is only one 
> contact registered.  If there is more than one contact registered, the INVITE 
> is sent to the registered contact and then the PBX doesn't know how to route 
> the call since it hits a default contact.  Is there a better way to do this?  
> I'm fairly new to Opensips and would appreciate any help. :-)
>
> A snippet from my script:
> route {
>        ...
>        if (!lookup("location","m")) {
>                switch ($retcode) {
>                        case -1:
>                        case -3:
>                                t_newtran();
>                                t_reply("404", "Not Found");
>                                exit;
>                        case -2:
>                                sl_send_reply("405", "Method Not Allowed");
>                                exit;
>                }
>        }
>
>        $rU=$oU;
>        route(1);
> }
>
>
> route[1] {
>        if (is_method("INVITE")) {
>                t_on_branch("2");
>                t_on_reply("2");
>                t_on_failure("1");
>        }
>
>        if (!t_relay()) {
>                sl_reply_error();
>        };
>        exit;
> }
>
>
> branch_route[2] {
>        xlog("new branch at $ru\n");
> }
>
>
> onreply_route[2] {
>        xlog("incoming reply\n");
> }
>
>
> failure_route[1] {
>        if (t_was_cancelled()) {
>                exit;
>        }
>
>        if (t_check_status("3[0-9][0-9]")) {
>        t_reply("404","Not found");
>                exit;
>        }
> }
>
>
>
> Thanks,
> Spencer
>
>
>
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Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber

[Dave Singer]

Toyima,
Not sure how much you know about branches, so...

With that switch branches are added for each alias. By default any
test or action in the rest of the main route script only applies to
the first branch. That is why only the first branch has the lookup
location done to it.
In order to test/act on other branches, you need to use t_on_branch()
and define a route block. When t_on_branch() is set to that route
block it will be run for all branches that were created.
Also of note is that in failure route, the vars like $ru and headers
are all set to the way they were before any branch route is run (may
have an effect on how you failover to voicemail). So consider that
when choosing to apply changes in main routing or branch routing. A
good example is uac_replace_from() which can only be run once on a
message so it is good to put it as one of the last things and if there
are branches, do it in the branch block.
What I've found to work good is to define a named route block to
handle last minute changes and test in main route block if there are
multiple branches. If only main branch, just call the
last_minute_changes from main route before exit; other wise set
t_on_branch() and call last_minute_changes from the branch route
block.

Hope that helps.

Dave

On Thu, May 19, 2011 at 2:58 AM, Toyima Dias  wrote:
> Hi Dave,
>
> i've added the line modparam("alias_db", "append_branches", 1), but now i'm
> trying to call the alias 314001 and the INVITE is going to both users 1000
> and 1001 as expected: that's nice...
>
>  #
> U 2011/05/18 13:55:52.918314 192.168.131.129:5060 -> 192.168.131.1:61052
> INVITE sip:vipexamk@192.168.131.1:61052 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.0
> Via: SIP/2.0/UDP
> 192.168.131.1:5060;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z-
> Max-Forwards: 69
> Contact: 
> To: 
> From: "1000";tag=ff69d424
> Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Content-Type: application/sdp
> User-Agent: Zoiper rev.5324
> Content-Length: 329
>
> v=0
> o=Zoiper_user 0 0 IN IP4 192.168.131.1
> s=Zoiper_session
> c=IN IP4 192.168.131.1
> t=0 0
> m=audio 8000 RTP/AVP 3 0 8 110 98 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> #
> U 2011/05/18 13:55:52.918371 192.168.131.129:5060 -> 192.168.131.129:5060
> INVITE sip:1000@192.168.131.129 SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 192.168.131.129;branch=z9hG4bKec48.66d63466.1
> Via: SIP/2.0/UDP
> 192.168.131.1:5060;branch=z9hG4bK-d8754z-801d2084170f1b21-1---d8754z-
> Max-Forwards: 69
> Contact: 
> To: 
> From: "1000";tag=ff69d424
> Call-ID: N2JiMTRhODJhNTRiNTgzNjc3NjdkZTI1NThiOWQ1MTU.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Content-Type: application/sdp
> User-Agent: Zoiper rev.5324
> Content-Length: 329
>
> v=0
> o=Zoiper_user 0 0 IN IP4 192.168.131.1
> s=Zoiper_session
> c=IN IP4 192.168.131.1
> t=0 0
> m=audio 8000 RTP/AVP 3 0 8 110 98 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> As you can see, the R-URI of the first invite is ok, but not the second, as
> you can see the domain part is the ip address of the proxy and not the ip of
> the location user 1000, what's happening? seems to be a loop on the second
> branch (192.168.131.129:5060 -> 192.168.131.129:5060)...as i can see, the DB
> alias is correctly fetching both users from DB, but only the first one is
> resolved via lookup(location) (registration search).
>
> any suggest?
>
> Many thanks!
>
> 2011/5/18 Dave Singer 
>>
>> Toyima,
>>
>> check:
>> http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030
>>
>> Dave
>>
>> On Tue, May 17, 2011 at 6:44 AM, Toyima Dias  wrote:
>> >
>> > Is it possible to add the same alias to more than one subscriber? for
>> > example, if an incoming call to the number 14882736524 is assigned to
>> > subscriber 1000, is it possible to assign this DID to subscriber 1001? the
>> > idea is to create a group of ringing or something like that, is that
>> > possible? i've tried to insert the data on phpmyadmin and als

Re: [OpenSIPS-Users] adding 1 alias to more than one subscriber

[Dave Singer]

Toyima,

check:
http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030

Dave

On Tue, May 17, 2011 at 6:44 AM, Toyima Dias  wrote:
>
> Is it possible to add the same alias to more than one subscriber? for 
> example, if an incoming call to the number 14882736524 is assigned to 
> subscriber 1000, is it possible to assign this DID to subscriber 1001? the 
> idea is to create a group of ringing or something like that, is that 
> possible? i've tried to insert the data on phpmyadmin and also on 
> opensips-panel, it doesn't work:
>
>
> Error
>
> consulta SQL:
>
> INSERT INTO `opensips`.`dbaliases` (
>
> `id` ,
> `alias_username` ,
> `alias_domain` ,
> `username` ,
> `domain`
> )
> VALUES (
> NULL , '314001', '192.168.131.129', '1000', '192.168.131.129'
> )
>
> MySQL ha dicho:
>
> #1062 - Duplicate entry '314001-192.168.131.129' for key 'alias_idx'
>
>
> Thanks!
>
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Re: [OpenSIPS-Users] opensips+rtpproxy strange locks

[Dave Singer]

Nick,

Have you verified there is no firewall on the opensips server with
   iptables -L
I believe the dump pulls the packets before they go through iptables.
So what you capture may be just dropped by iptables and not reaching
opensips.
Though really that shouldn't be a problem since iptables should be
keeping track of the conversation and allowing replies.
If there are active rules, try
   iptables -F
to see if no firewall makes a difference.

Dave

On Tue, May 17, 2011 at 8:11 AM, Razvan Crainea
 wrote:
> Hi Nick,
>
> Can you send me the tcpdump trace just when OpenSIPS sends the offer and
> waits for the reply?
>
> Regards,
> Razvan
>
>
>
> On 05/17/2011 05:23 PM, n...@uni-petrol.com wrote:
>>
>> Dear Razvan!
>>
>> Exactly!
>> And it is worst that opensips hang for 15-20 minutes and stop responding
>> on client requests (REGISTER/OPTIONS/INVITE/etc) and only write in log ERROR
>> messages.
>> If I restart opensips then it see response, begin response on client
>> request and write that rtpproxy found:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
>> for it enabled
>>
>> On Tue, 17 May 2011 16:06:22 +0300, Razvan Crainea wrote:
>>
>>> Hi Nick,
>>> So you can see RTPProxy's reply with tcpdump, but OpenSIPS doesn't and
>>> signals that error?
>>>
>>> On 05/17/2011 03:24 PM, n...@uni-petrol.com [12]wrote:
>>>
 Dear Razvan! I don't have firewall between rtpproxy and opensips at all
 so this is not a problem. rtpproxy are on same LAN and in the same
 broadcast. Also I run tcpdump when this errors appear and I see packets
 in both direction. Regarding downgrading rtpproxy, I just replace
 rtpproxy binary without changing anything. On Tue, 17 May 2011 14:39:07
 +0300, Razvan Crainea wrote:

> Hello Nick, It seems that RTPProxy responds to your queries, but
> OpenSIPS doesn't receive them properly. So probably there is somebody
> in the middle blocking the messages, most likely a firewall. Have you
> changed the socket when tried with the older version? Regards, Razvan
> On 05/17/2011 02:01 PM, n...@uni-petrol.com [9][9]wrote:
>
>> Dear Razvan! Unfortunately my mail client strip some chars. The
>> real string was:May 13 17:10:51 /usr/sbin/opensips[7990]:
>> ERROR:rtpproxy:send_rtpp_command: proxydoes not respond, disable it
>> I think there is some bug in rtpproxy, because I downgrade rtpproxy
>> to older version with ag-project patches from opensips.org site and
>> have not such locks anymore. On Tue, 17 May 2011 11:23:17 +0300,
>> Razvan Crainea wrote:
>>
>>> Hello Nick, Have you removed the RTPProxy IP from the logs, or
>>> these are the exactly error logs that OpenSIPS provide? Regards,
>>> Razvan On 05/13/2011 07:35 AM, n...@uni-petrol.com
>>> [3][3][3]wrote:
>>>
 Dear All! opensips + rtpproxy latest from trunk. I catch
 strange opensips locks. Opensips stop responding and output to
 syslog only these errors: May 10 13:21:47
 /usr/sbin/opensips[3662]: ERROR:rtpproxy:send_rtpp_command:
 proxy :2> does not respond, disable it May 10 13:21:48
 /usr/sbin/opensips[3661]: ERROR:rtpproxy:send_rtpp_command:
 proxy :2> does not respond, disable it May 10 13:22:05
 /usr/sbin/opensips[3660]: ERROR:rtpproxy:send_rtpp_command:
 proxy :2> does not respond, disable it ... May 10 13:29:22
 /usr/sbin/opensips[5428]: ERROR:rtpproxy:send_rtpp_command:
 proxy :2> does not respond, disable it I tried increase
 debug level through fifo but verbosity don't increase. In
 rtpproxy in debug mode I have these messages:
 DBUG:handle_command: received command "3660_595
 Uc8,0,18,4,3,110,97,101 0e70ed1b54b0727d20297cdf7da7ae33@:5060
 12290 as6c1cbf1f;1" INFO:handle_command: new session
 0e70ed1b54b0727d20297cdf7da7ae33@:5060, tag as6c1cbf1f;1
 requested, type strong INFO:handle_command: new session on a
 port 13184 created, tag as6c1cbf1f;1 INFO:handle_command:
 pre-filling caller's address with :12290 DBUG:doreply: sending
 reply "3660_595 13184 " DBUG:handle_command: received command
 "3660_596 Uc0,8,18,3,101 0e70ed1b54b0727d20297cdf7da7ae33@:5060
 16496 as1a95022b;1 as6c1cbf1f;1" INFO:handle_command: adding
 strong flag to existing session, new=1/0/0 INFO:handle_command:
 lookup on ports 13184/18324, session timer restarted
 INFO:handle_command: pre-filling callee's address with :16496
 DBUG:doreply: sending reply "3660_596 18324 "
 DBUG:handle_command: received command "3662_610 Uc0,8,18,3,101
 0e70ed1b54b0727d20297cdf7da7ae33@:5060 16496 as1a95022b;1
 as6c1cbf1f;1" INFO:handle_command: adding strong flag to
 existing session, new=1/0/0 INFO:handle_command: lookup on
 ports 13184/18324, session timer restarted DBUG:doreply:
 sending reply "3662_61

Re: [OpenSIPS-Users] FIFO very fragile

[Dave Singer]

Brett,

In your perl app are you starting and forking/threading the read before
   print FIFO ":uptime:reply_fifo\n";
?
I seem to remember starting the read process first to be very important.
My speculation is the reply fifo is not truely active until it has a
process actively attached and reading from it. So when opensips tries
to write to it, it succeeds checking the fifo file exists and it has
proper properties but when it tries to write to it linux says it is
not ready or something since it is not a buffer per say. Then opensips
waits for something to update/push it which never happens. (again just
my speculation from my previous efforts)

Dave

On Mon, May 9, 2011 at 5:01 PM, Brett Nemeroff  wrote:
>
> Dave,
> Yeah, I do actually get a reply. Works reliably too as long as I'm careful 
> about what I'm asking for. I'm trying to do all of this directly from perl. 
> For some reason I can't write to the fifo with a simple:
> open(FIFO,">/tmp/opensips_fifo");
> print FIFO ":uptime:reply_fifo\n";
> The fifo never receives it.. now, if I leave the reader attached and use 
> "echo" from bash, it works great.
> Really, the concern I have it doing things that cause the fifo to "die" and 
> become unavailable. I've seen this with the opensipsctl script as well, where 
> the fifo just stops responding until you restart.
> -Brett
>
> On Mon, May 9, 2011 at 4:03 PM, Dave Singer  wrote:
>>
>> Are you actually able to get results out of the reply fifo?
>> It's been a long time since I created my bash fifo script so I don't 
>> remember all the particulars of what I ran into.
>> It can be important to start the read fifo before sending the command
>>
>> Hear is a snippet from a bash fifo I did.
>>
>> trap "\rm -f $dlg $fifo_reply ${fifo_answer}* 2> /dev/null; exit 1" 0
>> if [ ! -w $FIFO ] ; then # can I write to FIFO server?
>>     echo "Error opening ser's FIFO $FIFO" >> $DEBUG_OUT
>>     echo "FAILED"
>>     exit 1
>> fi
>> mkdir -p $fifo_dir 2> /dev/null
>> mkfifo $fifo_reply # create a reply FIFO
>> if [ $? -ne 0 ] ; then
>>     echo "error opening reply fifo $fifo_reply" >> $DEBUG_OUT
>>     echo "FAILED"
>>     exit 1
>> fi
>> chmod a+w $fifo_reply
>> # start reader now so that it is ready for replies
>> # immediately after a request is out
>> cat < $fifo_reply > $dlg  &
>> fifo_job="$!"
>>
>> # set trap to cleanup in case of problems or outside kill
>> trap "kill -9 $fifo_job $fifo_answer_job 2>1 >> $DEBUG_OUT ;  exit 1" 0
>> # finally actually push command to the fifo
>> echo "reply pid: $fifo_job, reply fifo: $fifo_reply" >> $DEBUG_OUT
>>
>> You may want to just use the fifo handeling built into opensipsctl
>> eg:
>> /path/opensipsctl fifo ps
>>
>> Dave
>>
>>
>> On Mon, May 9, 2011 at 1:42 PM, Brett Nemeroff  wrote:
>>>
>>> Dave,
>>> I don't think those are the issues. First of all, if I supply the full path 
>>> to the reply fifo, I get an error message that the filename is invalid. 
>>> More importantly, it doesn't work. :) without the full path, it does work.
>>> Secondly, I don't think it's SELinux issue because it *does work* 
>>> repeatedly over and over. But an extra carriage return spoils the fun for 
>>> everyone and I can't fix it without restarting opensips.
>>> Do you believe SELinux could cause an issue like that?
>>> Thanks!
>>> -Brett
>>>
>>> On Mon, May 9, 2011 at 3:39 PM, Dave Singer  
>>> wrote:
>>>>
>>>> Brett,
>>>> I believe you need the full path to the reply fifo.
>>>> Try:
>>>> echo -e ":address_dump:/tmp/my_fifo\n\n" > /tmp/opensips_fifo
>>>>
>>>> you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux 
>>>> issue (very likely if SELinux is enabled, I posted a howto for SELinux in 
>>>> the list a number of months back)
>>>>
>>>> Dave
>>>>
>>>> On Mon, May 9, 2011 at 9:40 AM, Brett Nemeroff  wrote:
>>>>>
>>>>> Hello List,
>>>>> So I've been doing some testing with the mi_fifo and found that it 
>>>>> appears to be really fragile. Here's what I've done:
>>>>> mkfifo /tmp/my_fifo
>>>>> cat /tmp/my_fifo&
>>>>> e

Re: [OpenSIPS-Users] Pjsip dialler with openIMS core

[Dave Singer]

Looks like you sent your request to the wrong list.

On Mon, May 9, 2011 at 12:24 AM, abid khan  wrote:

> Hello everybody,
>   I want to know main criteria or parameters which
> i have to fulfill if i want to register* pjsip dialer* with openIMS core.
> When i try to register pjsip dialler with openims core my server send does
> not send "Authorization " to pjsip in response and then 504 (request time
> out) in end. So any one can please help me in this.
>
> --
> *Abid khan
>  *
>
> ___
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>
>
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Re: [OpenSIPS-Users] FIFO very fragile

[Dave Singer]

Are you actually able to get results out of the reply fifo?
It's been a long time since I created my bash fifo script so I don't
remember all the particulars of what I ran into.
It can be important to start the read fifo before sending the command

Hear is a snippet from a bash fifo I did.

trap "\rm -f $dlg $fifo_reply ${fifo_answer}* 2> /dev/null; exit 1" 0
if [ ! -w $FIFO ] ; then # can I write to FIFO server?
echo "Error opening ser's FIFO $FIFO" >> $DEBUG_OUT
echo "FAILED"
exit 1
fi
mkdir -p $fifo_dir 2> /dev/null
mkfifo $fifo_reply # create a reply FIFO
if [ $? -ne 0 ] ; then
echo "error opening reply fifo $fifo_reply" >> $DEBUG_OUT
echo "FAILED"
exit 1
fi
chmod a+w $fifo_reply
# start reader now so that it is ready for replies
# immediately after a request is out
cat < $fifo_reply > $dlg  &
fifo_job="$!"

# set trap to cleanup in case of problems or outside kill
trap "kill -9 $fifo_job $fifo_answer_job 2>1 >> $DEBUG_OUT ;  exit 1" 0
# finally actually push command to the fifo
echo "reply pid: $fifo_job, reply fifo: $fifo_reply" >> $DEBUG_OUT

You may want to just use the fifo handeling built into opensipsctl
eg:
/path/opensipsctl fifo ps

Dave


On Mon, May 9, 2011 at 1:42 PM, Brett Nemeroff  wrote:

> Dave,
> I don't think those are the issues. First of all, if I supply the full path
> to the reply fifo, I get an error message that the filename is invalid. More
> importantly, it doesn't work. :) without the full path, it does work.
>
> Secondly, I don't think it's SELinux issue because it *does work*
> repeatedly over and over. But an extra carriage return spoils the fun for
> everyone and I can't fix it without restarting opensips.
>
> Do you believe SELinux could cause an issue like that?
> Thanks!
> -Brett
>
>
> On Mon, May 9, 2011 at 3:39 PM, Dave Singer wrote:
>
>> Brett,
>> I believe you need the full path to the reply fifo.
>> Try:
>> echo -e ":address_dump:/tmp/my_fifo\n\n" > /tmp/opensips_fifo
>>
>> you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux
>> issue (very likely if SELinux is enabled, I posted a howto for SELinux in
>> the list a number of months back)
>>
>> Dave
>>
>> On Mon, May 9, 2011 at 9:40 AM, Brett Nemeroff wrote:
>>
>>> Hello List,
>>> So I've been doing some testing with the mi_fifo and found that it
>>> appears to be really fragile. Here's what I've done:
>>>
>>> mkfifo /tmp/my_fifo
>>> cat /tmp/my_fifo&
>>> echo -e ":address_dump:my_fifo\n\n" > /tmp/opensips_fifo
>>> *returns*
>>> 200 OK
>>>   48 <1.2.3.4,2, 0, 0, NULL, NULL>
>>> ** cat process ends
>>> cat /tmp/my_fifo&
>>> echo -e ":address_dump:my_fifo\n\n" > /tmp/opensips_fifo
>>> ** nothing
>>>
>>> At this point, I can't make the fifo work again until I restart opensips.
>>> If I detach from the fifo (kill the cat, so to speak), and reattach it
>>> doesn't work. Nothing seems to make it responsive again. At first I though
>>> something was horribly broken, but then I removed one of my \n from the fifo
>>> command and now it works "as expected". Two new-lines shouldn't break the
>>> fifo, right?
>>>
>>> Thanks!
>>> -Brett
>>>
>>>
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>> David Singer
>>
>> ___
>> Users mailing list
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] FIFO very fragile

[Dave Singer]

Brett,
I believe you need the full path to the reply fifo.
Try:
echo -e ":address_dump:/tmp/my_fifo\n\n" > /tmp/opensips_fifo

you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux
issue (very likely if SELinux is enabled, I posted a howto for SELinux in
the list a number of months back)

Dave

On Mon, May 9, 2011 at 9:40 AM, Brett Nemeroff  wrote:

> Hello List,
> So I've been doing some testing with the mi_fifo and found that it appears
> to be really fragile. Here's what I've done:
>
> mkfifo /tmp/my_fifo
> cat /tmp/my_fifo&
> echo -e ":address_dump:my_fifo\n\n" > /tmp/opensips_fifo
> *returns*
> 200 OK
>   48 <1.2.3.4,2, 0, 0, NULL, NULL>
> ** cat process ends
> cat /tmp/my_fifo&
> echo -e ":address_dump:my_fifo\n\n" > /tmp/opensips_fifo
> ** nothing
>
> At this point, I can't make the fifo work again until I restart opensips.
> If I detach from the fifo (kill the cat, so to speak), and reattach it
> doesn't work. Nothing seems to make it responsive again. At first I though
> something was horribly broken, but then I removed one of my \n from the fifo
> command and now it works "as expected". Two new-lines shouldn't break the
> fifo, right?
>
> Thanks!
> -Brett
>
>
>
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


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Re: [OpenSIPS-Users] Validate To Uri

[Dave Singer]

Jan,

I'm assuming you actually meant $tU which is the user portion of the To URI.
If all you are trying to do is test if they are numeric then regex is the
best way to test it.
Something like:
if ( ! $tU =~ "^[0-9]*$" ) {
   ... deal with non numeric user ...
}

Dave

On Wed, May 4, 2011 at 2:07 AM, Denis Putyato  wrote:

> Hello
>
> Try to use dialplan module
> http://www.opensips.org/html/docs/modules/devel/dialplan.html#id249075
>
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] On Behalf Of Jan D.
> Sent: Wednesday, May 04, 2011 1:00 PM
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] Validate To Uri
>
> In an INVITE I want to validate the To Uri ($Tu). This should be a numeric
> value (ie. 31201234567).
>
> Is there a function I can use or should I validate against a regexp (if so,
> do you have an example).
>
> Regards,
>
> Jan
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Validate-To-Uri-tp6330114p6330114.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Mediaproxy remove unknown session errors

[Dave Singer]

Jeff,

Sounds like the bye is not getting matched with the dialog. There are some
settings for dialog module for matching fallback that you probably want to
look at.
Also there are a number of threads that have discussed matching problems
that may be helpful.

Dave

On Tue, May 3, 2011 at 2:46 PM, Jeff Pyle  wrote:

> Hello,
>
> I use engage_media_proxy() in this case to call Mediaproxy 2.4.4 from
> Opensips 1.6.4.  Each day my syslog is full of messages like this:
>
> media-dispatcher[25715]: error: Got `remove' command from OpenSIPS for
> unknown session with call-id '[call-id]'.
>
>
> So far today I count 328 of them.  What would cause the dispatcher to
> seemly forget about a call that was set up with engage_media_proxy()?
>
>
> - Jeff
>
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Re: [OpenSIPS-Users] high process count

[Dave Singer]

Bret,

try:
ps aux --forest | grep opensips
that will show you parent and child processes in tree diagram and if any
opensips master process is actually responsible for so many of them.
I agree that it is probably being spawned many times with a cron job, loop
in the init script or a service like monit.

Dave

On Sat, Apr 30, 2011 at 12:53 PM, Bret McDanel wrote:

> On Sat, 2011-04-30 at 19:37 +0100, Stanisław Pitucha wrote:
> > You mean that fifo ps returns a low number of processes? How are you
> > starting up opensips? What kind of scripts / process management are
> > you using?
> >
> > I would guess that something you use to start up opensips is not
> > working as you expect and keeps respawning it all the time. Of course
> > new processes won't stay around for long because they'll try to bind
> > on some ports which are already taken. But if you're respawning often
> > enough, the total count could go into thousands.
> >
>
>
> The extra processes are because it is forking doing something, it is not
> multiple instances being started like you suggested.  They  all have the
> same parent ID.
>
> ps auxww | grep opensips | wc -l
> the list keeps growing.  And if I kill opensips most die but not all,
> and those are only terminated with a kill -9.
>
> I show about 7 with opensipsctl ps but hundreds (it constantly grows
> they never die).
>
> They are clearly blocking somewhere, and it cannot be a syscall because
> a kill -9 cant take those out.  I am about to dig in and try to find out
> where and more importantly why.
>
>
> I start opensips fork=no logstderr=yes debug=10 with only a -f and -P
> option in a standard shell.
>
> --
> Trixter aka Bret McDanel
> website:  http://www.0xdecafbad.com
> pgp key:  http://bit.ly/9XYK4b
>
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



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Re: [OpenSIPS-Users] Opensips geographic redundancy

[Dave Singer]

Marcello,

Might check into getting an IP with BGP (Border Gateway Protocol) or some
other IP routing protocol. Not all ISPs offer these services and it is
usually offered on a subnet of addresses not a single address. If your
providers/contacts can't do this or don't have good answers you might check
with www.spectrumnet.us.

Dave

On Fri, Apr 29, 2011 at 9:22 AM, Marcello Lupo wrote:

> Hi to all,
> i have a problem and hope someone can give me a clue on it.
> We have an Opensips infrastructure with heartbeat HA and it is working
> fine.
> I'm implementing the geographic redundancy on our VoIP infrastructure with
> opensips 1.6.4.
> There are 2 different locations. 2 server in each location (one running the
> Opensips and the other running the mysql DB).
> The 2 structures are totally independent but share the same DB data (with
> mysql master->slave->master replication). So each location can handle
> customer requests independently, the registrations are duplicated from one
> server to the other to maintain all the CPE's reachable from both servers.
> Now I'd like to use the DNS SRV records to let the CPEs to use the 2
> servers in a kind of load balance.
> I discovered that lot of CPE are not implementing the SRV records logic in
> a correct way.
>
> Patton CPE with 5.7 firmware make the first REGISTER (without
> authentication) on one server and after the server answer back with 401
> Unauthorized the CPE retry the REGISTER (with authentication) on the other
> server that reject it because an invalid nonce is found (it was generated
> from the other server obviously). It should continue the session with the
> same server it started with the first REGISTER because it received an answer
> and the server is alive.
>
> Asterisk 1.6 is making the REGISTER properly (one for each server) but
> place all the calls through only one server even if it is down.
>
> I'm sure will find that lot of other CPE will have trouble with SRV records
> correct implementation.
>
> I'd like to find a way to do it in a CPE software independent way.
> So i'm starting to search another solution that let me implement the
> geographic redundancy without the SRV records but I'm short of ideas now
> without inserting any Single Point of Failure in the system.
>
> I thought to a front end proxy (in HA redundancy like the one of now) to be
> a load balancer to the other 2 proxies but in any way i don't' have a
> geographic redundancy.
>
> Have you any suggestions?
> Thank you all.
> Regards,
> Marcello
>
>
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>



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Re: [OpenSIPS-Users] need OpenSIPS installation

[Dave Singer]

Jeff,

I would be interested in working with you. I've been integrating
OpenSIPS in hosted VOIP environment for 4 years. Unfortunately I will
be laid off the end of this week, Apr 22 (I'm the last one to go).
You can look at my resume on Linkedin (
http://www.linkedin.com/in/davesingercando ) or Monster.com or I can
send you one if you are interested.

Thanks,
Dave


On Wed, Apr 6, 2011 at 9:50 AM, Jeff Matson
 wrote:
> We are looking to implement OpenSIPS within our hosted VOIP environment.  If
> you are interested in contracting out to assist us in the configuration and
> initial programming, please let me know.
>
>
>
> regards
>
>
>
> Jeff
>
>
>
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

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[OpenSIPS-Users] Problem with Diversion when added in failure route and RURI uses DNS SRV

[Dave Singer]

I have a server running 1.6.3. In failure route I add a Diversion and
the RURI is a DNS SRV record and sometimes one of the servers times
out and opensips auto tries the second server from the SRV records but
without the diversion header in the new message
Is this proper/expected behavior or a bug?

I would think for SRV it would only change the destination IP.
Especially where you don't have an opportunity to make any fixes to
the original message (what it was just before any branches).

Dave

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Re: [OpenSIPS-Users] Need ideas to tamper with CSeq

[Dave Singer]

Cinthia,

The CSeq needs to be changed before it hits opensips. So you could do
something where if the cancel is not matched, lookup the right cseq
using a memcache lookup (that you stored during the accepted invite)
and send it out to it self with the modified CSeq. I wouldn't be
surprised if this too doesn't work since it would be coming from
itself rather than the ATA like the INVITE. Might be worth a try
though.

Dave

On Wed, Mar 30, 2011 at 5:21 PM, Cindy Leung  wrote:
> I know I'm doing something bad here.  However, we are having a problem with 
> one of the SIP phones that we support.  When it sends out an INVITE and then 
> CANCEL, the CANCEL is not being forwarded.  We are suspecting that it is 
> caused by a wrong CSeq value.
>
> INVITE #1 gets challenged.
> INVITE #2 accepted.
> CANCEL is sent, but CSeq is the same as the one in INVITE #1.
>
> It is less than ideal for us to contact their support and we'd like to get it 
> fixed asap.  I've tried subst(), remove_hf and append_hf to play with CSeq 
> with no luck.
>
> Any suggestions?  Thanks!
>
>
> Cinthia
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Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

Duong,

That is probably a selinux problem.
I dealt with that a couple months ago and posted a little howto for
easily dealing with selinux problems the right way (not disabling
selinux).
Just google for:
selinux opensips dave

Dave

On Thu, Mar 24, 2011 at 8:57 PM, Duong Manh Truong
 wrote:
> Hi Dave!
> I changed my config to
> "$box_id=0;
> // mi host:port pair || fifo_file
> //$boxes[$box_id]['mi']['type']="fifo";
> //$boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";
> $boxes[$box_id]['mi']['conn']="127.0.0.1:8000";
> $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";
> "
> And i've got the same results: OK with add users only!
> When i hit the "apply changes to server" button, i see this
> "sorry -- cannot open write fifo"
> Or when using MI commands:
> "10:40:18  |   opensipsctl online   |  /tmp/opensips_fifo
> 
>
> sorry -- cannot open write fifo"
>
> Check the file:
> [root@opensips tmp]# ls -l opensips_fifo
> prw-rw-rw-. 1 root root 0 Mar 25 10:31 opensips_fifo
> I think that file's permission here is ok !!
> Continued checking the opensips.cfg, i had added the lines following
> "
> # - mi_fifo params -
> modparam("mi_fifo", "fifo_mode", 0666)
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> "
> But error is still occured :-(
> My "acc table" has "cdr_id filed", not the "cdr field"
> additionally, i have "Cdrs" table separately.
> Both 2 tables above are empty. (although i made several internal calls)
> And i've checked the "location table" and it truly contained records of
> online users
> (not the presence module as you suggested)
> Please tell me what are the next steps to check ?
> Thanks!
>
>
> 2011/3/25 Dave Singer 
>>
>> Duong,
>> Sorry. I forgot I modified my CP and those options.
>> You need to have it:
>> $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";
>>
>> and remove:
>> $boxes[$box_id]['mi']['type']="fifo";
>> $boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";
>>
>> Are you sure cdrs are being recorded (you have seen cdrs in the acc
>> table)?
>> Have you verified online users is using the location (registered
>> users) table and not the presence module fifo/table (I haven't used
>> presence so don't know what is available and how CP would pull
>> presence info).
>> My guess is that online users uses presence module.
>>
>> Dave
>>
>> On Thu, Mar 24, 2011 at 11:31 AM, Duong Manh Truong
>>  wrote:
>> > Hi Dave,
>> > My default config in was :
>> > // default: $boxes[$box_id]['mi']['conn']="127.0.0.1:8000";
>> > // default: $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";
>> > then i changed it to
>> > $boxes[$box_id]['mi']['type']="fifo";
>> > $boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";
>> > And i found some functions of CP works such as: Add/edit users; Add/edit
>> > domain.
>> > But a lot of others still dont work such as: cdrs, show online users,
>> > 
>> > I see the "apply changes to server" button, i hit it and get this
>> > message
>> > for a new web browser tab:
>> > "http://10.2.14.122/cp/tools/system/domains/apply_changes.php
>> > Cannot connect to OpenSIPS Server via Management Interface ()"
>> > I come to /cp/tools/system/domains but find no "apply_changes.php" file
>> > :-(
>> > Please help me to find out how i can connect to Opensips Server and
>> > making
>> > changes
>> > Thanks!
>> > 2011/3/24 Dave Singer 
>> >>
>> >> Duong,
>> >>
>> >> The MI commands are for opensips and ps does NOT take any arguments.
>> >> The easiest way for it to connect when on the same box is with fifo.
>> >> Here is a snippet from the boxes.global.inc.php:
>> >> /* DEFINITION OF BOXES (servers)
>> >> */
>> >> // each server is a box
>> >> $box_id=0;
>> >> $boxes[$box_id]['mi']['type']="fifo";
>> >> $boxes[$box_id]['mi'

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

Duong,
Sorry. I forgot I modified my CP and those options.
You need to have it:
$boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";

and remove:
$boxes[$box_id]['mi']['type']="fifo";
$boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";

Are you sure cdrs are being recorded (you have seen cdrs in the acc table)?
Have you verified online users is using the location (registered
users) table and not the presence module fifo/table (I haven't used
presence so don't know what is available and how CP would pull
presence info).
My guess is that online users uses presence module.

Dave

On Thu, Mar 24, 2011 at 11:31 AM, Duong Manh Truong
 wrote:
> Hi Dave,
> My default config in was :
> // default: $boxes[$box_id]['mi']['conn']="127.0.0.1:8000";
> // default: $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";
> then i changed it to
> $boxes[$box_id]['mi']['type']="fifo";
> $boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";
> And i found some functions of CP works such as: Add/edit users; Add/edit
> domain.
> But a lot of others still dont work such as: cdrs, show online users,
> 
> I see the "apply changes to server" button, i hit it and get this message
> for a new web browser tab:
> "http://10.2.14.122/cp/tools/system/domains/apply_changes.php
> Cannot connect to OpenSIPS Server via Management Interface ()"
> I come to /cp/tools/system/domains but find no "apply_changes.php" file :-(
> Please help me to find out how i can connect to Opensips Server and making
> changes
> Thanks!
> 2011/3/24 Dave Singer 
>>
>> Duong,
>>
>> The MI commands are for opensips and ps does NOT take any arguments.
>> The easiest way for it to connect when on the same box is with fifo.
>> Here is a snippet from the boxes.global.inc.php:
>> /* DEFINITION OF BOXES (servers)
>> */
>> // each server is a box
>> $box_id=0;
>> $boxes[$box_id]['mi']['type']="fifo";
>> $boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";
>>
>> This does require that opensips module mi_fifo is loaded and
>> configured with the same path/file as you set in the config above
>> (above path is the default path).
>> opensipsctl uses it also (unless it was configured to use xmlrpc (not
>> sure it can)).
>> Opensips xmlrpc module by default lisetns on port 8000, not the sip
>> port 5060. That is why you got that message.
>> I've not been able to get xmlrpc to compile and work on Centos 5 servers.
>>
>> Dave
>>
>> On Thu, Mar 24, 2011 at 12:17 AM, Duong Manh Truong
>>  wrote:
>> >
>> > Thank Dave!
>> > When going to the "MI commands" tab, surprisingly i found that
>> > "Cannot connect to OpenSIPS Server via Management Interface
>> > (127.0.0.1/8000)
>> > Execute MI Command"
>> >
>> > After i executed the "ps -A" command, i only got this
>> > "14:02:20  |  ps -A   |  127.0.0.1:8000"
>> > The same result for "opensipsctl online" command
>> >  "14:04:37  |   opensipsctl online   |  127.0.0.1:8000"
>> > And i think the problem is my CP does not connect to my Opensips server
>> > on 5060 port. I remember that i have not configured CP listening port
>> > before, it is default :-(
>> > (CP & Opensips server were installed in the same machine)
>> > Checking these files "db.inc.php" ; local.inc.php; globals.php  in the
>> > CP config directory "/var/www/opensips-cp/config"
>> > I don't see anywhere to config CP listening port !
>> > Please help me to figure out, thanks!
>> > 2011/3/24 Dave Singer 
>> >>
>> >> Duong,
>> >>
>> >> Yea missed that. :-)
>> >> To check basic CP connection to opensips, go to System, MI Commands.
>> >> In box under Execute MI Command, put "ps" and click "execute". It
>> >> should show a bunch of lines about the opensips processes that are
>> >> running.
>> >>
>> >> As far as showing Online Users, it may be using presence module/table
>> >> rather than active registrations to qualify a user as "online".
>> >> Someone else will have to answer that or you can look at the CP source
>> >> code for that answer.
>> >>
>> &g

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

Duong,

The MI commands are for opensips and ps does NOT take any arguments.
The easiest way for it to connect when on the same box is with fifo.
Here is a snippet from the boxes.global.inc.php:
/* DEFINITION OF BOXES (servers) */
// each server is a box
$box_id=0;
$boxes[$box_id]['mi']['type']="fifo";
$boxes[$box_id]['mi']['fifo']="/tmp/opensips_fifo";

This does require that opensips module mi_fifo is loaded and
configured with the same path/file as you set in the config above
(above path is the default path).
opensipsctl uses it also (unless it was configured to use xmlrpc (not
sure it can)).
Opensips xmlrpc module by default lisetns on port 8000, not the sip
port 5060. That is why you got that message.
I've not been able to get xmlrpc to compile and work on Centos 5 servers.

Dave

On Thu, Mar 24, 2011 at 12:17 AM, Duong Manh Truong
 wrote:
>
> Thank Dave!
> When going to the "MI commands" tab, surprisingly i found that
> "Cannot connect to OpenSIPS Server via Management Interface (127.0.0.1/8000)
> Execute MI Command"
>
> After i executed the "ps -A" command, i only got this
> "14:02:20  |  ps -A   |  127.0.0.1:8000"
> The same result for "opensipsctl online" command
>  "14:04:37  |   opensipsctl online   |  127.0.0.1:8000"
> And i think the problem is my CP does not connect to my Opensips server on 
> 5060 port. I remember that i have not configured CP listening port before, it 
> is default :-(
> (CP & Opensips server were installed in the same machine)
> Checking these files "db.inc.php" ; local.inc.php; globals.php  in the CP 
> config directory "/var/www/opensips-cp/config"
> I don't see anywhere to config CP listening port !
> Please help me to figure out, thanks!
> 2011/3/24 Dave Singer 
>>
>> Duong,
>>
>> Yea missed that. :-)
>> To check basic CP connection to opensips, go to System, MI Commands.
>> In box under Execute MI Command, put "ps" and click "execute". It
>> should show a bunch of lines about the opensips processes that are
>> running.
>>
>> As far as showing Online Users, it may be using presence module/table
>> rather than active registrations to qualify a user as "online".
>> Someone else will have to answer that or you can look at the CP source
>> code for that answer.
>>
>> Dave
>>
>> On Wed, Mar 23, 2011 at 12:35 PM, Duong Manh Truong
>>  wrote:
>> > Thank David,
>> > But i found that in my opensips.cfg file:
>> > "
>> > # - usrloc params -
>> > # modparam("usrloc", "db_mode",   0)
>> > /* uncomment the following lines if you want to enable DB persistency
>> >    for location entries */
>> > modparam("usrloc", "db_mode",   2)
>> > modparam("usrloc", "db_url",
>> > "mysql://opensips:opensipsrw@localhost/opensips")
>> > "
>> > So i think that my "db_mode" is not the cause of my failed CP
>> > " 2 - Write-Back scheme. This is a combination of previous two schemes. All
>> > changes are made to memory and database synchronization is done in the
>> > timer. The timer deletes all expired contacts and flushes all modified or
>> > new contacts to database. Use this scheme if you encounter high-load peaks
>> > and want them to process as fast as possible. The mode will not help at all
>> > if the load is high all the time. Also, latency of this mode is much lower
>> > than latency of mode 1, but slightly higher than latency of mode 0."
>> > My CP still does  not work :-( , even the simple function like "show online
>> > user"  --- "no data found"
>> > (While checking by using opensipsctl is ok)
>> >  opensipsctl online
>> > database engine 'MYSQL' loaded
>> > Control engine 'FIFO' loaded
>> > 1001
>> >
>> > Please tell me how to check my CP and its connection to Opensips server?
>> > Thanks !
>> >
>> >
>> >
>> >
>> > --
>> >
>> > Message: 1
>> > Date: Tue, 22 Mar 2011 21:06:21 -0700
>> > From: Dave Singer 
>> > Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
>> >        effect into Opensips Server
>> > To: OpenSIPS users mailling list 
>> > Message-ID:
>> >        
>> > Cont

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

Duong,

Yea missed that. :-)
To check basic CP connection to opensips, go to System, MI Commands.
In box under Execute MI Command, put "ps" and click "execute". It
should show a bunch of lines about the opensips processes that are
running.

As far as showing Online Users, it may be using presence module/table
rather than active registrations to qualify a user as "online".
Someone else will have to answer that or you can look at the CP source
code for that answer.

Dave

On Wed, Mar 23, 2011 at 12:35 PM, Duong Manh Truong
 wrote:
> Thank David,
> But i found that in my opensips.cfg file:
> "
> # - usrloc params -
> # modparam("usrloc", "db_mode",   0)
> /* uncomment the following lines if you want to enable DB persistency
>    for location entries */
> modparam("usrloc", "db_mode",   2)
> modparam("usrloc", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips")
> "
> So i think that my "db_mode" is not the cause of my failed CP
> " 2 - Write-Back scheme. This is a combination of previous two schemes. All
> changes are made to memory and database synchronization is done in the
> timer. The timer deletes all expired contacts and flushes all modified or
> new contacts to database. Use this scheme if you encounter high-load peaks
> and want them to process as fast as possible. The mode will not help at all
> if the load is high all the time. Also, latency of this mode is much lower
> than latency of mode 1, but slightly higher than latency of mode 0."
> My CP still does  not work :-( , even the simple function like "show online
> user"  --- "no data found"
> (While checking by using opensipsctl is ok)
>  opensipsctl online
> database engine 'MYSQL' loaded
> Control engine 'FIFO' loaded
> 1001
>
> Please tell me how to check my CP and its connection to Opensips server?
> Thanks !
>
>
>
>
> --
>
> Message: 1
> Date: Tue, 22 Mar 2011 21:06:21 -0700
> From: Dave Singer 
> Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
>        effect into Opensips Server
> To: OpenSIPS users mailling list 
> Message-ID:
>        
> Content-Type: text/plain; charset=ISO-8859-1
>
> The key I think is usrloc db mode is 0, which means db is not used.
> http://www.opensips.org/html/docs/modules/1.6.x/usrloc.html#id292029
>
> On Tue, Mar 22, 2011 at 6:47 PM, Duong Manh Truong
>  wrote:
>> Dear Dave!
>> Thanks for your help.
>> I've checked the opensips.cfg again but i still can not find out what
>> stuff
>> is incorrect ! :-(
>> I attached my config file in this email, hope that will help my problem to
>> be?clearer!
>> If anyone has had the same issue, please tell me how can i check or fix
>> it.
>> (Hope to receive relatively specific informations)
>> Thanks and Best Regards.
>>
>>
>>
>> Date: Mon, 21 Mar 2011 08:45:03 -0700
>> From: Dave Singer 
>> Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
>> ? ? ? ?effect into Opensips Server
>> To: OpenSIPS users mailling list 
>> Message-ID:
>> ? ? ? ?
>> Content-Type: text/plain; charset=ISO-8859-1
>>
>> Duong,
>>
>> Some modules, like drouting, you have to click a reload module button
>> in the control pannel.
>> However, I think your problem is that you haven't implemented the
>> needed stuff in opensips.cfg script.
>> CP only provides an interface to managing/accessing the databases
>> tables. You still have to impliment the logic in opensips.cfg.
>>
>> Dave
>>
>> On Sat, Mar 19, 2011 at 3:04 AM, Duong Manh Truong
>>  wrote:
>>> Hi all,
>>> I have followed all instructions about Opensips CP installations
>>> After done all configurations, the Web interface is appeared
>>> but unfortunately, when i use modules in this interfaces, it seems there
>>> is
>>> no connection between Opensips CP and Opensips Server Core!
>>> Take add/manage users module as an example: I registered user 1001, then
>>> i
>>> selected this module to show online users
>>> => But the result was "no data found" :-(
>>> I've created opensips database, correctly insert parameters to opensips
>>> CP
>>> config files (including mysql user/pass to connect to opensips database)
>>> No error found !
>>> But i still can not use Opensips CP, someone help me to figure out why
>>> Opensips CP has no effect ?

Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

The key I think is usrloc db mode is 0, which means db is not used.
http://www.opensips.org/html/docs/modules/1.6.x/usrloc.html#id292029

On Tue, Mar 22, 2011 at 6:47 PM, Duong Manh Truong
 wrote:
> Dear Dave!
> Thanks for your help.
> I've checked the opensips.cfg again but i still can not find out what stuff
> is incorrect ! :-(
> I attached my config file in this email, hope that will help my problem to
> be clearer!
> If anyone has had the same issue, please tell me how can i check or fix it.
> (Hope to receive relatively specific informations)
> Thanks and Best Regards.
>
>
>
> Date: Mon, 21 Mar 2011 08:45:03 -0700
> From: Dave Singer 
> Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
>        effect into Opensips Server
> To: OpenSIPS users mailling list 
> Message-ID:
>        
> Content-Type: text/plain; charset=ISO-8859-1
>
> Duong,
>
> Some modules, like drouting, you have to click a reload module button
> in the control pannel.
> However, I think your problem is that you haven't implemented the
> needed stuff in opensips.cfg script.
> CP only provides an interface to managing/accessing the databases
> tables. You still have to impliment the logic in opensips.cfg.
>
> Dave
>
> On Sat, Mar 19, 2011 at 3:04 AM, Duong Manh Truong
>  wrote:
>> Hi all,
>> I have followed all instructions about Opensips CP installations
>> After done all configurations, the Web interface is appeared
>> but unfortunately, when i use modules in this interfaces, it seems there
>> is
>> no connection between Opensips CP and Opensips Server Core!
>> Take add/manage users module as an example: I registered user 1001, then i
>> selected this module to show online users
>> => But the result was "no data found" :-(
>> I've created opensips database, correctly insert parameters to opensips CP
>> config files (including mysql user/pass to connect to opensips database)
>> No error found !
>> But i still can not use Opensips CP, someone help me to figure out why
>> Opensips CP has no effect ??
>> Do i misconfig ?(or forget to config) some important parameters to make
>> Opensips CP connect properly to Opensips Server Core?
>> (Something likes authenticationsetc)
>> Thanks a lot!
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
>
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Re: [OpenSIPS-Users] Opensips Control Pannel does not take effect into Opensips Server

[Dave Singer]

Duong,

Some modules, like drouting, you have to click a reload module button
in the control pannel.
However, I think your problem is that you haven't implemented the
needed stuff in opensips.cfg script.
CP only provides an interface to managing/accessing the databases
tables. You still have to impliment the logic in opensips.cfg.

Dave

On Sat, Mar 19, 2011 at 3:04 AM, Duong Manh Truong
 wrote:
> Hi all,
> I have followed all instructions about Opensips CP installations
> After done all configurations, the Web interface is appeared
> but unfortunately, when i use modules in this interfaces, it seems there is
> no connection between Opensips CP and Opensips Server Core!
> Take add/manage users module as an example: I registered user 1001, then i
> selected this module to show online users
> => But the result was "no data found" :-(
> I've created opensips database, correctly insert parameters to opensips CP
> config files (including mysql user/pass to connect to opensips database)
> No error found !
> But i still can not use Opensips CP, someone help me to figure out why
> Opensips CP has no effect ??
> Do i misconfig  (or forget to config) some important parameters to make
> Opensips CP connect properly to Opensips Server Core?
> (Something likes authenticationsetc)
> Thanks a lot!
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Asynchronous DB queries in OpenSIPS 1.x

[Dave Singer]

Vlad,
Would it be possible extend DB_VITRUAL kind of a combination of your
two options and Brett's suggestion.
I'm just guessing how things might work and possibilities. So for what
it is worth...
Since it is a middle layer, it could either use the async capabilities
of the native driver or put it in a queue to be handled by a special
db queue process that could also be threaded potentially (if
beneficial). The IPC would only need to pass the query that is needed.
To take advantage of the async the user would only need to use the
DB_VIRTUAL module.
Probably would want a module param to enable/disable async
(async_db_updates) in case it is important and maybe a bit flag option
as well that would reverse the async_db_updates option when flag set
for case by case basis.
Potentially with a separate queue process updates like registrations
could be combined into a single update.

Dave


On Wed, Mar 16, 2011 at 10:32 AM, Brett Nemeroff  wrote:
>
>
>
>
> On Wed, Mar 16, 2011 at 12:18 PM, Vlad Paiu  wrote:
>>
>> Hello all,
>>
>> Problem :
>> 1) Extend the OpenSIPS DB core. Add extra core processes that would only
>> handle queries that return no results.
>>     For example : The accounting module need to insert an entry in the DB.
>> The module calls the insert() function. Behind the scene, this triggers
>> passing all the arguments to the new core processes, via IPC mechanisms. The
>> insert() then exists and the SIP children continues execution as if the
>> entry has been inserted in the DB. Meanwhile, the DB core processes receive
>> the new parameters, build and send the query, blocking if necessary.
>>
>
> Maybe I'm just saying the same thing another way, but what about an async
> execution queue. So you basically add to the queue messages to be executed
> to the database and on some sort of timer loop process them. To the script,
> we just assume everything is 100% successful.
> Is IPC really necessary for this? The goal here is really just to offload
> the processing elsewhere so that the DB slowness doesn't adversely affect
> opensips core performance. right?
>
> I thin it's also important that there is a async execute and a sync execute
> and people (users?) need to know that in the async execute, you won't know
> if the execute succeeded in the script logic *ever*
> -Brett
>
>
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Re: [OpenSIPS-Users] t_relay(tcp:)

[Dave Singer]

Jeff,

I like using the core rewritable variables to control things. They
allow you to easily change just one part of the request and many times
the equivalent function does not accept variables as arguments.

$rP RURI protocol: http://www.opensips.org/Resources/DocsCoreVar16#toc65
$fs Forced socket: http://www.opensips.org/Resources/DocsCoreVar16#toc38

Dave


On Mon, Mar 7, 2011 at 5:31 AM, Jeff Chua  wrote:
>
> t_relay() default to udp. How can force it to tcp without specifying the host?
>
>
> Thanks,
> Jeff
>
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Re: [OpenSIPS-Users] Stopping a T.38 reinvite with Opensips loose_route()

[Dave Singer]

JP,

The two end points have to agree on an RTP protocol.
You may look at the SDP in the initial INVITEs to see if T.38 is offered and
if it is, remove it.
Also if the re-INVITE contains G711 along with the T.38 you should be able
to just remove the T.38.
That should accomplish the same effect as the customer disabling T.38.

FYI: FaxBack has a very reliable T.38 implementation NET SatisFAXtion that
we have been using for quite some time.

Dave

On Sun, Mar 6, 2011 at 1:52 PM, Iñaki Baz Castillo  wrote:

> 2011/3/6 Jeff Pyle :
> > Unfortunately the customer cannot disable T.38.  I'd like to be able to
> > reject the reinvite with a send_reply("488", "Not acceptable here")
> within
> > the loose_route() section.  Is this legit, or am I going to break things,
> > like getting the two endpoints' CSeqs out of sync?
>
> You are not going to break CSeq at all. The proxy is able to reject
> in-dialog requests depending on local policy. When the UA receives 488
> it does know that, in case of a new in-dialog request, it must use a
> greater CSeq, which of course will be valid in the other UA (it must
> be just higher than previous one, no matter how much higher).
>
> But in your case, the problem is that when your UA receives the 488 it
> will probably send a BYE (it's the usual policy when a re-INVITE is
> rejected). But you can try it. You will not break SIP protocol at all.
>
> --
> Iñaki Baz Castillo
> 
>
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Re: [OpenSIPS-Users] How much A records are retained in IP comparison

[Dave Singer]

Igor,

I'm pretty sure that won't work at all. You would need to use the
transformation ip.resolve.
I don't know if it returns multiple IPs but if it does you would want to
assign it to an avp that can handle multiple.
An example to try:
$var(fqdn) = "some.server.com"; # Not sure if you can do a transformation
directly on a string. Never tried.
$avp(s:ips) = $(var(fqdn){ip.resolve});
xlog("INFO", "$(avp(s:ips)[0])::$(avp(s:ips)[1])::$(avp(s:ips)[2])\n");
if ( avp_check("$avp(s:ips)", "eq/$si/") ) {     }

Let me know what you find.
Dave

On Wed, Mar 2, 2011 at 9:20 AM, Igor Solovyov  wrote:

> Hi,
>
> If I use
> if ($si=="some.server.com")
> {
> ...
> }
> and some.server.com has several A records,
> are all of them used in upper comparison?
> How does TTL influence on saved IPs?
>
> Regards,
> Igor.
>
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Re: [OpenSIPS-Users] route based on previous load_balance dialog

[Dave Singer]

stefano,

You might look at the dialog module. Check if what you need is stored in the
db for a dialog. You would need to have the db mode set to realtime.
The nice thing if you can use dialog module is that it is built in to write
the info to the db and clean it up.
Otherwise you will need to write to a custom table and cleanup it up at the
end of the call.
Either way you would use avp_db_query to pull the needed info from the db.

Dave

On Fri, Feb 25, 2011 at 8:34 AM, Stefano Sasso  wrote:

> Hello folks,
>  I have this load_balance situation:
> opensips in front of 3 asterisk servers in a outbound only call-center.
>
> When a agent places a call, opensips load balance through the three
> asterisks.
> Now, we need to use the "ExtensSpy" command from asterisk, dialing
> something like ***123 (where 123 is the extension to spy). So,
> opensips need to rewritehostport directly to the asterisk server which
> is handling 123's conversation, without using load_balance.
>
> I thought to get informations from the dialog database table, in fact
> a row is like this:
> caller_contact: sip:8...@aa.cc.dd.82:1029;transport=UDP
> callee_contact: sip:xx...@aa.bb.cc.54
>
> where 8002 is the caller extension and aa.bb.cc.54 is the asterisk
> server involved in the conversation.
>
> but... how can I get and use this information from opensips configuration?
>
> thanks so much!
>
> bye,
> stefano
>
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Re: [OpenSIPS-Users] running opensips under another user

[Dave Singer]

Anton,

When you are trying to start it are you starting it using the -u opensips
and -g opensips command line options to have it run as opensips user and
group?
That way in the startup script you can easily remove the dead pid as root
(checking first to be sure opensips is not actually running).

Dave

On Thu, Feb 24, 2011 at 11:55 PM, Anton Zagorskiy <
a.zagors...@oyster-telecom.ru> wrote:

> Hi, Dave.
>
> The problem was I’v started openSIPS under not opensips user. With ‘sudo –u
> opensips /usr/local/sbin/opensips …’ openSIPS starts well.
>
>
>
> But now it seems that pid file isn’t erased when openSIPS failed to start.
>
>
>
>
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Dave Singer
> *Sent:* Friday, February 25, 2011 4:14 AM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] running opensips under another user
>
>
>
> Anton,
>
> Have you checked the selinux logs?
>
> Search the list about selinux. A couple months back I posted what I ran
> into and how to solve it fairly simply.
>
>
>
> Dave.
>
> On Thu, Feb 24, 2011 at 6:23 AM, Duane Larson 
> wrote:
>
> And you're positive that the /var/run/opensips directory is owned by the
> opensips:opensips user and group?  All I ever do is
>
>
>
> sudo mkdir /var/run/opensips/
> sudo chown opensips:opensips /var/run/opensips
>
>
>
> Never had that issue.
>
>
>
> cd /var/run
>
> ls -la
>
> drwxr-xr-x  2 opensipsopensips4096 Feb 24 08:23 opensips
>
>
>
>
>
>
>
> On Thu, Feb 24, 2011 at 3:07 AM, Anton Zagorskiy <
> a.zagors...@oyster-telecom.ru> wrote:
>
> Hi.
>
> I've made a user opensips with group opensips and with shell nologin.
> Also, I've created /var/run/opensips with opensips::opensips and access
> mode
> 775
>
> Next, running openSIPS with -u opensips -g opensips -P
> /var/run/opensips/opensips.pid
> And opensips failed to start with:
>
> ERROR:core:daemonize: unable to create pid file
> /var/run/opensips/opensips.pid: Permission denied
>
>
> A command "sudo -u opensips touch /var/run/opensips/opensips1.pid" works
> well.
>
> What I'm doing wrong?
>
>
>
> WBR, Anton Zagorskiy
> VoIP Developer, Oyster Telecom
> Phone.: +7 812 601-0666
> Fax: +7 812 601-0593
> a.zagors...@oyster-telecom.ru
> www.oyster-telecom.ru
>
>
>
>
>
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>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>
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>
>
>
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Re: [OpenSIPS-Users] running opensips under another user

[Dave Singer]

Anton,
Have you checked the selinux logs?
Search the list about selinux. A couple months back I posted what I ran into
and how to solve it fairly simply.

Dave.

On Thu, Feb 24, 2011 at 6:23 AM, Duane Larson wrote:

> And you're positive that the /var/run/opensips directory is owned by the
> opensips:opensips user and group?  All I ever do is
>
> sudo mkdir /var/run/opensips/
> sudo chown opensips:opensips /var/run/opensips
>
> Never had that issue.
>
> cd /var/run
> ls -la
> drwxr-xr-x  2 opensipsopensips4096 Feb 24 08:23 opensips
>
>
>
>
> On Thu, Feb 24, 2011 at 3:07 AM, Anton Zagorskiy <
> a.zagors...@oyster-telecom.ru> wrote:
>
>> Hi.
>>
>> I've made a user opensips with group opensips and with shell nologin.
>> Also, I've created /var/run/opensips with opensips::opensips and access
>> mode
>> 775
>>
>> Next, running openSIPS with -u opensips -g opensips -P
>> /var/run/opensips/opensips.pid
>> And opensips failed to start with:
>>
>> ERROR:core:daemonize: unable to create pid file
>> /var/run/opensips/opensips.pid: Permission denied
>>
>>
>> A command "sudo -u opensips touch /var/run/opensips/opensips1.pid" works
>> well.
>>
>> What I'm doing wrong?
>>
>>
>>
>> WBR, Anton Zagorskiy
>> VoIP Developer, Oyster Telecom
>> Phone.: +7 812 601-0666
>> Fax: +7 812 601-0593
>> a.zagors...@oyster-telecom.ru
>> www.oyster-telecom.ru
>>
>>
>>
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>
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>
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Re: [OpenSIPS-Users] {ip.resolve} script transformation trouble

[Dave Singer]

Igor,

Only documentation I see it in is for development
version<http://www.opensips.org/Resources/DocsCoreTran#toc47> implying
trunk what will be 1.6.5. However a little while back I was looking at the
source code of 1.6.4 for transformations and noticed some IP transformation
stuff. I can't tell you ip.resolve is there but it is a place to start.

Dave

On Thu, Feb 24, 2011 at 1:39 PM, Igor Solovyov  wrote:

> Thanks Dave for prompt help.
>
> Now I get
>
> Feb 24 23:32:32 os1 opensips: ERROR:core:parse_transformation: unknown
> transformation: [ip] in [{ip.resolve}]
> Feb 24 23:32:32 os1 opensips: ERROR:core:parse_transformation: error
> parsing [{ip.resolve}]
> Feb 24 23:32:32 os1 opensips: ERROR:core:pv_parse_spec: ERROR:bad tr in
> pvar name "var"
> Feb 24 23:32:32 os1 opensips: ERROR:core:pv_parse_spec: invalid parsing in
> [$(var(fs_addr_s){ip.resolve})] at (4)
> Feb 24 23:32:32 os1 opensips: CRITICAL:core:yyerror: parse error in config
> file, line 144, column 49-78: unknown script variable
>
> which leads me to though that my OpenSIPS version 1.6.3 doesn't support
> "ip" transformation.
> Am I correct? And so, from which version "ip" is supported?
> Or may be is there another method to get resolved ip by string?
>
> Igor.
>
>
> On Thu, Feb 24, 2011 at 11:06 PM, Dave Singer 
> wrote:
>
>> Igor,
>>
>> Very close. Notice the ( between $ and var and ) after var name:
>>
>> $var(fs_addr) = $(var(fs_addr_s){ip.resolve}{ip.ntop});
>>
>> see syntax near top of http://www.opensips.org/Resources/DocsCoreVar16
>>
>> Dave
>>
>> On Thu, Feb 24, 2011 at 10:36 AM, Igor Solovyov wrote:
>>
>>> Hi All,
>>>
>>> I try to execute in the script
>>> $var(fs_addr_s) = "opensips.org"; # just for test
>>> $var(fs_addr) = $var(fs_addr_s{ip.resolve}{ip.ntop});
>>> xlog("L_ERR", "fs_addr = $var(fs_addr)");
>>> but in log I can see only 0 instead of resolved ip address.
>>> What I do wrongly?
>>>
>>> Thanks,
>>> Igor.
>>>
>>>
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>>>
>>
>
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Re: [OpenSIPS-Users] {ip.resolve} script transformation trouble

[Dave Singer]

Igor,

Very close. Notice the ( between $ and var and ) after var name:

$var(fs_addr) = $(var(fs_addr_s){ip.resolve}{ip.ntop});

see syntax near top of http://www.opensips.org/Resources/DocsCoreVar16

Dave

On Thu, Feb 24, 2011 at 10:36 AM, Igor Solovyov  wrote:

> Hi All,
>
> I try to execute in the script
> $var(fs_addr_s) = "opensips.org"; # just for test
> $var(fs_addr) = $var(fs_addr_s{ip.resolve}{ip.ntop});
> xlog("L_ERR", "fs_addr = $var(fs_addr)");
> but in log I can see only 0 instead of resolved ip address.
> What I do wrongly?
>
> Thanks,
> Igor.
>
>
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Re: [OpenSIPS-Users] pseudo-variable problem with increasing cps

[Dave Singer]

Ronald,

The only time I've seen it be null in failure route is if there was no reply
received,
you might add a check to see if it was a local timeout:

if ( t_local_replied("all") ) {
   xlog("did not get any response");
} else {
  xlog("$(ci): $C(rx)failure route: $(rs)
$(rr)$C(xx)\n");
}

Dave

On Wed, Feb 23, 2011 at 10:04 AM, Ronald Cepres  wrote:

> Hi all,
>
> I'm setting up opensips as a stateful proxy, and i have the following
> snippet of code on the failure route:
>
> failure_route[1] {
> xlog("$(ci): $C(rx)failure route: $(rs)
> $(rr)$C(xx)\n");
>
> # Failure route routine...
> }
>
> The values of the call-id, response code and reason are normally there.
>
> but on higher cps, i often get this:
>
> : failure route:  
>
> Does this mean that the transaction was already destroyed before opensips
> got the reply?
>
> I wonder if anyone comes across this problem. Any help would be
> appreciated. Thanks!
>
>
>
> Regards,
>
> Ronald
>
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Re: [OpenSIPS-Users] Modify a URI

[Dave Singer]

Or just:

$rU = "12345#" + $rU;

Dave

On Wed, Feb 23, 2011 at 10:36 AM, Max Mühlbronner  wrote:

>  Hello,
>
>
>
> if you just want to add a prefix, you could use prefix() which is very easy
> to use. J
>
>
>
> Regards
>
>
>
> Max M.
>
>
>
> *Von:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *Im Auftrag von *Brian Artigas
> *Gesendet:* Mittwoch, 23. Februar 2011 19:31
> *An:* users@lists.opensips.org
> *Betreff:* [OpenSIPS-Users] Modify a URI
>
>
>
> Can anyone give me a code example how I would take a URI and inject a
> string between the sip: and the actual number. I found the rewriteuri()
> function but cannot find any code examples on how to use it.
>
>
>
> For instance take the request URI:   sip:5615551212@111.222.333.444
>
>
>
> And change it to:
> sip:12345#5615551212@111.222.333.444
>
>
>
> Thanks in advance,
>
> Brian
>
>
>
>
>
> Eingehende eMail ist virenfrei.
> Von AVG überprüft - www.avg.de
> Version: 9.0.872 / Virendatenbank: 271.1.1/3463 - Ausgabedatum: 02/23/11
> 12:32:00
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[OpenSIPS-Users] Dialog module app callback hooks not saved to DB

[Dave Singer]

I tested setting up acc module to use the cdr_flag with dialog module.
Works nicely until opensips is restarted while there is an open call.
After an opensips restart, calls that were started before the restart do not
get an entry in the DB. Not even an old style BYE record.
I verified that the dialog was matched using:
  xlog("L_NOTICE", "CID->$ci, fU->$fU, orU->$oU: Loose Route $rm Request -
si=$si - next via $rd:$rp\n");
if ( $DLG_status == NULL ) {
xlog("L_WARN", "CID->$ci, fU->$fU, orU->$oU: WARNING: BYE without
dialog. NO CDR, just old style rec for BYE in acc table.\n");
in the loose routing logic that handles the bye.
I get the first message but not the warning.
I also ran into this problem previously with media_proxy_engage. It cleans
itself up because of no RTP after a bit though and so is not as big of a
deal.

Am I missing something or is this a bug I need to report?

Dave
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Re: [OpenSIPS-Users] non-transactional 4XX messages

[Dave Singer]

Jason,
That is very strange behavior. If there is no matching transaction I
think there is no way to catch the message as reply and failure routes
are triggered by tm module.
I'm not sure since I haven't setup a stateless config.

I'm curious too if there is something that can be done here.

Dave

On Mon, Feb 21, 2011 at 7:08 PM, Tyler Merritt  wrote:
> You know what I do?
> I add append_hf("GW: BLAH") and change BLAH for each step in the if blocks.
> Then you can grab a tcpdump and see your custom header.  This has helped me
> ENORMOUSLY during testing to track packet flow.
> I prefer looking at the packets themselves and the custom headers rather
> than searching through the logs.
> It's a bit time consuming to add all the append_hf lines in nearly every if
> () block, but man, after a while I started "seeing" the code :)
>
>
> On Tue, Feb 22, 2011 at 11:54 AM, Jason Yeh  wrote:
>>
>> Tyler,
>> Yes, I have already thought the same thing. However that 403 SIP Message
>> didn't land there in the main routing block, yet through pcap traces I could
>> see the SIP Message being routed to the end-user client. I had this basic
>> log of every new messages routed to the main route, for example:
>> 
>> route {
>>
>> # Simple Log
>>
>> xlog("L_INFO", " M=$rm RURI=$ru F=$fu USR=$tU T=$tu IP=$si ID=$ci\n");
>>
>> 
>> 
>> I don't see 403 message landed there.
>> I tried to add logic of something like this on main route:
>> 
>>
>> # Detect if SIP/2.0 403 Message
>> if ( status == "403" ) {
>>
>> xlog("got SIP 403");
>>
>> }
>>
>> 
>>
>> Unfortunately, realized that the "status" reference of current message
>> would only work for the onreply route I'm sure there is reason for it,
>> most likely developers might can answer that.
>> -Jason
>>
>> On Mon, Feb 21, 2011 at 9:10 PM, Tyler Merritt  wrote:
>>>
>>> I would do this by matching the method of the packet in the main routing
>>> block, and then using textops to search for the 403 in a subsequent if () {}
>>> block, and then just drop; or exit;
>>> My methods are usually not the most elegant solution - I'll wait for Dave
>>> or one of the Devs to chime in with a better method :)
>>>
>>> On Tue, Feb 22, 2011 at 10:09 AM, Jason Yeh  wrote:

 Hello everyone,
 This certain SIP UAS keeps on sending "SIP/2.0 403" messages to my
 OpenSIPS machine out of blue, this failure/warning messages has nothing to
 do with this certain client on the call. I wanted to figure out how to drop
 ("not proxy") that annoying non-transactional 4XX SIP Messages being routed
 through this proxy server, however I couldn't figure out where to get this
 message landed to my routing script. Because the fact that this requests
 wasn't initially sent by proxy, it will not land to the "failure_route"
 routing block.
 In summary, how can I configure OpenSIPS routing logic to handle the
 non-transactional 4XX request sent by someone else out of blue?
 Thanks,
 Jason
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

>>>
>>
>>
>>
>> --
>> Regards,
>>
>> Jason Yeh
>> Co-Founder & CEO
>> Streamoso, LLC.
>>
>> This e-mail, including any attachments may contain information that is
>> protected by law as PRIVILEGED AND CONFIDENTIAL and is intended solely for
>> the use of the recipient or the employee or agent responsible for delivering
>> the message to the recipient. Please note that if you are not the intended
>> recipient, you are hereby notified that any dissemination, copying,
>> distribution, retention, re-transmission, printing or any other use of this
>> e-mail or the information contained herein is strictly prohibited. If you
>> have received this e-mail communication in error, please immediately send an
>> e-mail reply to notify the sender and immediately and permanently delete
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>
>
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Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

[Dave Singer]

Toyima,

Try:
yum whatprovides '*/md5' '*/md5sum' '*/db_dump'
the db_dump should be apparent from that.
I'm using Centos and it shows several packages that provide a md5
command but none put it in the path. You could try installing one and
linking it into the path to see if that helps.
 eg.: ln -s /var/lib/imap/md5 /usr/local/bin/
However I'm guessing two things.
  1. md5 is not critical as it kept going and stopped after another error.
  2. it is not actually wanting md5 but md5sum (provided by package
coreutils), the commonly installed program, and the reference needs to
be fixed in Osipsconsole. It looks like md5 is a library that some
applications include like mail apps and script languages. It also
might be that on different platforms, what is named md5sum on ours is
named md5 on others like maybe bsd or sun unix for example.

I don't know what you are trying to use the console for but for the
limited things I need to do with command line stuff I just use the
opensipsctl like .../opensipsctl fifo ps.
For db stuff I use opensips control pannel web project or just direct
sql commands.

Dave

On Mon, Feb 21, 2011 at 1:17 AM, Toyima Dias  wrote:
>
> Hi Dave,
>
> This is what i did:
>
> root@OpenSIPS home# yum search md5
> Loaded plugins: rhnplugin, security
> 
>  Matched: md5 
> =
> cyrus-sasl-md5.i386 : CRAM-MD5 and DIGEST-MD5 support for Cyrus SASL.
> perl-Digest-HMAC.noarch : Digest-HMAC Perl module
> Then:
>
> root@OpenSIPS home# yum install perl-Digest-HMAC.noarch
> And still not working...i'm working on Red Hat; have found the same problem 
> on google 
> (http://lists.opensips.org/pipermail/users/2010-November/015448.html) to 
> another person and seems that he never solve the situation. Any help will be 
> appreciated.
>
> 2011/2/18 Dave Singer 
>>
>> Toyima,
>> It is looking for those executable files. It looks like it first tries 
>> db_dump then because that fails tries db4.4_dump.
>> So you need to install the packages that supply md5 and either db_dump 
>> or db4.4_dump.
>> Dave
>>
>> On Fri, Feb 18, 2011 at 3:20 AM, Toyima Dias  wrote:
>>>
>>> Hi Dave,
>>>
>>> 2011/2/17 Dave Singer 
>>>>
>>>> Toyima,
>>>>
>>>> Have you configured it to connect to your database?
>>>
>>>
>>> the file opensipsctlrc is already configured to connect to my database 
>>> where i'm storing domains, locations, subscribers, etc etc...the DB_PATH 
>>> parameter commented and i don't now how to configure this parameter :S
>>>
>>>>
>>>> Have you installed packages that provide the commands it was
>>>> complaining about missing? From what I have seen, db is almost always
>>>> installed and it may just require it to be there even if it is not
>>>> used.
>>>
>>>
>>> The messages it is complaining are the following:
>>>
>>> which: no md5 in 
>>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>>> which: no db_dump in 
>>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>>> which: no db4.4_dump in 
>>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>>>
>>> Do i have to install md5, db_dump and db4.4_dump?
>>>
>>>>
>>>> Dave
>>>>
>>>> On Thu, Feb 17, 2011 at 1:50 AM, Toyima Dias  wrote:
>>>> >
>>>> > Hello Dave,
>>>> >
>>>> > 2011/2/17 Dave Singer 
>>>> >>
>>>> >> Toyima,
>>>> >>
>>>> >> On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias  wrote:
>>>> >> >
>>>> >> >
>>>> >> > 2011/2/16 Dave Singer 
>>>> >> >>
>>>> >> >> Toyima,
>>>> >> >> This is the right list. You just have to take into account that no 
>>>> >> >> one
>>>> >> >> is getting paid to answer your questions in the list. And sometimes
>>>> >> >> that means waiting a couple days.
>>>> >> >> Best tactic I've seen to keep putting it in front

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

[Dave Singer]

Toyima,

It is looking for those executable files. It looks like it first tries
db_dump then because that fails tries db4.4_dump.
So you need to install the packages that supply md5 and either db_dump or
db4.4_dump.

Dave

On Fri, Feb 18, 2011 at 3:20 AM, Toyima Dias  wrote:

> Hi Dave,
>
> 2011/2/17 Dave Singer 
>
>> Toyima,
>>
>> Have you configured it to connect to your database?
>>
>
> the file opensipsctlrc is already configured to connect to my database
> where i'm storing domains, locations, subscribers, etc etc...the DB_PATH
> parameter commented and i don't now how to configure this parameter :S
>
>
>> Have you installed packages that provide the commands it was
>> complaining about missing? From what I have seen, db is almost always
>> installed and it may just require it to be there even if it is not
>> used.
>>
>
> The messages it is complaining are the following:
>
> which: no md5 in
> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
> which: no db_dump in
> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
> which: no db4.4_dump in
> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>
> Do i have to install md5, db_dump and db4.4_dump?
>
>
>>
>> Dave
>>
>> On Thu, Feb 17, 2011 at 1:50 AM, Toyima Dias  wrote:
>> >
>> > Hello Dave,
>> >
>> > 2011/2/17 Dave Singer 
>> >>
>> >> Toyima,
>> >>
>> >> On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias 
>> wrote:
>> >> >
>> >> >
>> >> > 2011/2/16 Dave Singer 
>> >> >>
>> >> >> Toyima,
>> >> >> This is the right list. You just have to take into account that no
>> one
>> >> >> is getting paid to answer your questions in the list. And sometimes
>> >> >> that means waiting a couple days.
>> >> >> Best tactic I've seen to keep putting it in front of people is to
>> >> >> respond to your own thread with a little update of what you've tried
>> >> >> and the results.
>> >> >>
>> >> >
>> >> > Ok.
>> >> >
>> >> >>
>> >> >> Looks like it is looking for some tools: md5 and db ( I think
>> berkley
>> >> >> db ) tools.
>> >> >> Do you have it configured to connect to your DB 
>> >> >> etc/opensips/osipsconsolerc?
>> >> >
>> >> > Do you mean the configuration of /etc/opensips/opensipsctlrc? if so,
>> i've
>> >> > commented the following line:
>> >> Yes
>> >>
>> >> >
>> >> > # database path used by dbtext or db_berkeley
>> >> > #DB_PATH="/usr/local/etc/opensips/dbtext"
>> >> >
>> >> > Should i change something?
>> >> Did your results change at all after commenting it?
>> >>
>> >
>> > It was commented by default...i've uncommented and the same result my
>> friend
>> >
>> >>
>> >> The db connection info in that file should be the same as in your
>> >> opensips.cfg script, just different format for specifying it.
>> >>
>> >> Dave.
>> >> >
>> >> >>
>> >> >> I'd say configure it if not already, try again and see what it still
>> >> >> wants.
>> >> >> Then install packages that seem like what it is complaining about
>> not
>> >> >> finding. centos example search: yum whatprovides "*/db_dump"
>> >> >>
>> >> >> Dave
>> >> >>
>> >> >> On Tue, Feb 15, 2011 at 5:47 AM, Toyima Dias 
>> wrote:
>> >> >> > Is there any help or mailing list with the use of osipsconsole?
>> may
>> >> >> > be i'm
>> >> >> > not asking to the right mailing list :S
>> >> >> >
>> >> >> > 2011/2/15 Toyima Dias 
>> >> >> >>
>> >> >> >> Hello,
>> >> >> >> I'm having problems with my opensipsconsole, first, every time i
>> run
>> >> >> >> osipsconsole 

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

[Dave Singer]

Toyima,

Have you configured it to connect to your database?
Have you installed packages that provide the commands it was
complaining about missing? From what I have seen, db is almost always
installed and it may just require it to be there even if it is not
used.

Dave

On Thu, Feb 17, 2011 at 1:50 AM, Toyima Dias  wrote:
>
> Hello Dave,
>
> 2011/2/17 Dave Singer 
>>
>> Toyima,
>>
>> On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias  wrote:
>> >
>> >
>> > 2011/2/16 Dave Singer 
>> >>
>> >> Toyima,
>> >> This is the right list. You just have to take into account that no one
>> >> is getting paid to answer your questions in the list. And sometimes
>> >> that means waiting a couple days.
>> >> Best tactic I've seen to keep putting it in front of people is to
>> >> respond to your own thread with a little update of what you've tried
>> >> and the results.
>> >>
>> >
>> > Ok.
>> >
>> >>
>> >> Looks like it is looking for some tools: md5 and db ( I think berkley
>> >> db ) tools.
>> >> Do you have it configured to connect to your DB 
>> >> etc/opensips/osipsconsolerc?
>> >
>> > Do you mean the configuration of /etc/opensips/opensipsctlrc? if so, i've
>> > commented the following line:
>> Yes
>>
>> >
>> > # database path used by dbtext or db_berkeley
>> > #DB_PATH="/usr/local/etc/opensips/dbtext"
>> >
>> > Should i change something?
>> Did your results change at all after commenting it?
>>
>
> It was commented by default...i've uncommented and the same result my friend
>
>>
>> The db connection info in that file should be the same as in your
>> opensips.cfg script, just different format for specifying it.
>>
>> Dave.
>> >
>> >>
>> >> I'd say configure it if not already, try again and see what it still
>> >> wants.
>> >> Then install packages that seem like what it is complaining about not
>> >> finding. centos example search: yum whatprovides "*/db_dump"
>> >>
>> >> Dave
>> >>
>> >> On Tue, Feb 15, 2011 at 5:47 AM, Toyima Dias  wrote:
>> >> > Is there any help or mailing list with the use of osipsconsole? may
>> >> > be i'm
>> >> > not asking to the right mailing list :S
>> >> >
>> >> > 2011/2/15 Toyima Dias 
>> >> >>
>> >> >> Hello,
>> >> >> I'm having problems with my opensipsconsole, first, every time i run
>> >> >> osipsconsole on the my shell i receive some weird messages, take a
>> >> >> look:
>> >> >>
>> >> >> OpenSIPS:~#osipsconsole
>> >> >> which: no md5 in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db_dump in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db4.4_dump in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db4.5_dump in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db4.6_dump in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db_load in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db4.4_load in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> >> which: no db4.5_load in
>> >> >>
>> >> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin

Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

[Dave Singer]

Toyima,

On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias  wrote:
>
>
> 2011/2/16 Dave Singer 
>>
>> Toyima,
>> This is the right list. You just have to take into account that no one
>> is getting paid to answer your questions in the list. And sometimes
>> that means waiting a couple days.
>> Best tactic I've seen to keep putting it in front of people is to
>> respond to your own thread with a little update of what you've tried
>> and the results.
>>
>
> Ok.
>
>>
>> Looks like it is looking for some tools: md5 and db ( I think berkley
>> db ) tools.
>> Do you have it configured to connect to your DB 
>> etc/opensips/osipsconsolerc?
>
> Do you mean the configuration of /etc/opensips/opensipsctlrc? if so, i've
> commented the following line:
Yes

>
> # database path used by dbtext or db_berkeley
> #DB_PATH="/usr/local/etc/opensips/dbtext"
>
> Should i change something?
Did your results change at all after commenting it?

The db connection info in that file should be the same as in your
opensips.cfg script, just different format for specifying it.

Dave.
>
>>
>> I'd say configure it if not already, try again and see what it still
>> wants.
>> Then install packages that seem like what it is complaining about not
>> finding. centos example search: yum whatprovides "*/db_dump"
>>
>> Dave
>>
>> On Tue, Feb 15, 2011 at 5:47 AM, Toyima Dias  wrote:
>> > Is there any help or mailing list with the use of osipsconsole? may
>> > be i'm
>> > not asking to the right mailing list :S
>> >
>> > 2011/2/15 Toyima Dias 
>> >>
>> >> Hello,
>> >> I'm having problems with my opensipsconsole, first, every time i run
>> >> osipsconsole on the my shell i receive some weird messages, take a
>> >> look:
>> >>
>> >> OpenSIPS:~#osipsconsole
>> >> which: no md5 in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db_dump in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.4_dump in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.5_dump in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.6_dump in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db_load in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.4_load in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.5_load in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> which: no db4.6_load in
>> >>
>> >> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> >> Control engine FIFO loaded
>> >> OpenSIPS$:
>> >> As you can see, osipsconsole is "opened", but those messages are kind
>> >> of
>> >> wear, it shows the $PATH of root, why is this happening?
>> >>
>> >> And also, the biggest problem is that i can't add gateways, check the
>> >> following:
>> >>
>> >> OpenSIPS$:address add 0 192.168.1.192 255.255.255.255 5060 UDP
>> >> Used database is mysql
>> >> Warn fallback to local workdir ./mysql
>> >> Retrieving data from table failedYou have an error in your SQL syntax;
>> >> check the manual that corresponds to your MySQL server version for the
>> >> right
>> >> syntax to use near 'WHERE ip='192.168.1.192'' at line 1
>> >> Introducing the address rule 0 192.168.1.192 255.255.255.255 5060 UD

Re: [OpenSIPS-Users] Interproxy Authentication

[Dave Singer]

Juri,

If you are looking to get opensips1 itself to respond to a
proxy_challange of opensips2 your looking at a headache. Any way you
go you'll have to change the way things work to make a special case
for opensips1 on opensips2.
My suggestion would be to, where you are about to do the
proxy_authorize, check if is from opensips1 and if it is skip the
proxy_authorize.
Example:
if ( ! ( $si == "" || proxy_authorize("", "subscriber") ) ) {
 proxy_challenge("", "0");
 exit;
}
The source IP, $si, will be checked first and if it is a match, it
will short cut and proxy_authorize will not be called.
But if both tests fail a proxy_challenge is sent back.
If opensips1 is not at a static IP then you would probably want to
have opensips1 register, using uac module, with opensips2. On
opensips2 you would then do a db lookup to see if the source IP is the
same as what opensips registered from. That lookup could be cached so
you don't have to do a db call for every call.
If this is your case, search recent threads for cache_fetch for some
ideas of a similar use.

Dave

On Wed, Feb 16, 2011 at 1:36 AM, Juri Nysschen  wrote:
> Hi All,
>
>
>
> I want this scenario:
>
> UA - > Opensips1 -> Opensip2 -> PSTN
>
> Opensips2 has a set of subscribers and verifies authentication against a
> mysql db, keeps cdrs etc.
>
> Opensips1 has a different set of subscribers with db authentication and
> cdrs.
>
>
>
> The UA makes a call and is routed from Opensips1 to Opensips2 and then onto
> the PSTN gateway.
>
>
>
> My question is how do I impersonate the call made by UA as coming from a
> valid subscriber known to Opensips2?
>
>
>
> Currently I use asterisk to perform this role, but would ideally like to
> remove it from the chain:
>
> UA - > Opensips1 -> Asterisk -> Opensips2 -> PSTN
>
>
>
> Regards
>
> Juri Nysschen
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

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Re: [OpenSIPS-Users] Osipsconsole fail to add gateways

[Dave Singer]

Toyima,
This is the right list. You just have to take into account that no one
is getting paid to answer your questions in the list. And sometimes
that means waiting a couple days.
Best tactic I've seen to keep putting it in front of people is to
respond to your own thread with a little update of what you've tried
and the results.


Looks like it is looking for some tools: md5 and db ( I think berkley
db ) tools.
Do you have it configured to connect to your DB 
etc/opensips/osipsconsolerc?
I'd say configure it if not already, try again and see what it still wants.
Then install packages that seem like what it is complaining about not
finding. centos example search: yum whatprovides "*/db_dump"

Dave

On Tue, Feb 15, 2011 at 5:47 AM, Toyima Dias  wrote:
> Is there any help or mailing list with the use of osipsconsole? may be i'm
> not asking to the right mailing list :S
>
> 2011/2/15 Toyima Dias 
>>
>> Hello,
>> I'm having problems with my opensipsconsole, first, every time i run
>> osipsconsole on the my shell i receive some weird messages, take a look:
>>
>> OpenSIPS:~#osipsconsole
>> which: no md5 in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db_dump in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.4_dump in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.5_dump in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.6_dump in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db_load in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.4_load in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.5_load in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> which: no db4.6_load in
>> (/usr/kerberos/sbin:/usr/kerberos/bin:/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/usr/sbin:/sbin/:/usr/bin)
>> Control engine FIFO loaded
>> OpenSIPS$:
>> As you can see, osipsconsole is "opened", but those messages are kind of
>> wear, it shows the $PATH of root, why is this happening?
>>
>> And also, the biggest problem is that i can't add gateways, check the
>> following:
>>
>> OpenSIPS$:address add 0 192.168.1.192 255.255.255.255 5060 UDP
>> Used database is mysql
>> Warn fallback to local workdir ./mysql
>> Retrieving data from table failedYou have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the right
>> syntax to use near 'WHERE ip='192.168.1.192'' at line 1
>> Introducing the address rule 0 192.168.1.192 255.255.255.255 5060 UDP to
>> the database failed
>> You have an error in your SQL syntax; check the manual that corresponds to
>> your MySQL server version for the right syntax to use near '( grp, ip, mask,
>> port, proto , context_info ) VALUES ( 0, '192.168.1.192', '255.' at line 1
>> Execute 'address reload' to synchronize cache and database
>>
>> What should i check to sovle this situation?
>>
>> Best Regards!
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

[Dave Singer]

Congrats Juri!

That's great! Your welcome and thanks for sharing the working solution/hack. :-)
Now if you can just get them to fix their software. At least you have
a workaround while they do.

Dave

On Tue, Feb 15, 2011 at 10:32 PM, Juri Nysschen  wrote:
> Hi Dave,
>
> It worked! The CDRs are perfect. Thanks for the help.
>
> Here's the code in case anyone else has the same issue:
>
> rout{
>
>    ..
>
>    # handle cancel and re-transmissions
>        if (is_method("CANCEL") ) {
>                xlog("L_INFO","CANCEL [$fd/$fu/$rd/$ru/$si/]\n");
>                setflag(3); # start db accounting
>                setflag(5); # log failed transactions
>                route(5); # disconnect media proxy
>                if (t_check_trans()){
>                        t_relay();
>                        exit;
>                }
>                xlog("L_INFO","CANCEL Generated [$fd/$fu/$rd/$ru/$si/]\n");
>                # send a CANCEL downstream
>                exec_msg("/usr/sbin/opensipsctl fifo t_uac_cancel $ci $cs");
>
>                # send a ACK upstream for cdr purposes.
>                sl_send_reply("200","OK");
>                exit;
>        }
>
> }
>
> Regards
> Juri Nysschen
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
> Sent: Tuesday, February 15, 2011 8:28 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] FW: CANCELs with no transaction
>
> Juri,
>
> You should be able to get to it with
>  $(hdr(via){param.value,branch})
> But forcing it on the outbound cancel would probably be quite a trick
> since opensips wants to add It's via header automatically and that top
> via is what the next hop is counting on for matching the call.
> I think your best bet is to use the MI t_uac_cancel command as
> described last time. With it you don't have to same anything from the
> INVITE. You just use the callid and cseq from the CANCEL to pass to
> the MI command t_uac_cancel.
> Be aware you may have linux and selinux permissions conflicts for
> executing opensipsctl from inside opensips. I posted in a thread a
> thing about dealing with selinux in the last couple months that makes
> it quite easy if selinux is a problem and to see if it is the problem.
> Your command would be something like:
> exec("/usr/local/opensips/sbin/opensipsctl  fifo t_uac_cancel $ci $cs");
> Don't forget to make sure the source canceling the call gets the
> response they are looking for so they don't keep sending you cancels
> and you keep doing the exec over and over.
>
> Dave
>
> On Tue, Feb 15, 2011 at 5:52 AM, Juri Nysschen 
> wrote:
>> Hi Dave
>>
>> I have used wireshark to review the cancel message on a call timeout
> (which
>> works) and when received from the UA.
>>
>> The difference is in the branch= value as you predicted.
>>
>> How can I save the branch value on the INVITE and then change it on the
>> CANCEL?
>> I've tried all sorts of combinations for $branch but the vales are always
>> NULL
>>
>>
>> Regards
>> Juri Nysschen
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
>> Sent: Monday, February 14, 2011 10:16 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] FW: CANCELs with no transaction
>>
>> Juri,
>>
>> This was gnawing at me so I searched and found the "official" answer
>> of what a CANCEL should look like.
>> Go to
>> http://www.ietf.org/rfc/rfc3261.txt
>> and search for "9 Canceling a Request"
>> From what I read it basically says that the CANCEL should be nearly
>> identical to the INVITE it is canceling (without some headers that
>> aren't useful in a cancel like Supported)
>> It seems that the "branch=" , a "Magic Cookie", on the top most
>> via line is a turbo lookup to finding the transaction. That matches
>> the original INVITE and all is well. If it is not there then it has to
>> compare all the pertinent headers between invite and cancel.
>>
>> So if you can't get them to send the proper stuff, you could modify
>> the previous hack to store the branch=... that opensips put in the via
>> header it added (probably can't see it when sending the invite but it
>> is in the 1xx response to your invite.
>> Actually I just found one better I thin

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

[Dave Singer]

Juri,

You should be able to get to it with
  $(hdr(via){param.value,branch})
But forcing it on the outbound cancel would probably be quite a trick
since opensips wants to add It's via header automatically and that top
via is what the next hop is counting on for matching the call.
I think your best bet is to use the MI t_uac_cancel command as
described last time. With it you don't have to same anything from the
INVITE. You just use the callid and cseq from the CANCEL to pass to
the MI command t_uac_cancel.
Be aware you may have linux and selinux permissions conflicts for
executing opensipsctl from inside opensips. I posted in a thread a
thing about dealing with selinux in the last couple months that makes
it quite easy if selinux is a problem and to see if it is the problem.
Your command would be something like:
exec("/usr/local/opensips/sbin/opensipsctl  fifo t_uac_cancel $ci $cs");
Don't forget to make sure the source canceling the call gets the
response they are looking for so they don't keep sending you cancels
and you keep doing the exec over and over.

Dave

On Tue, Feb 15, 2011 at 5:52 AM, Juri Nysschen  wrote:
> Hi Dave
>
> I have used wireshark to review the cancel message on a call timeout (which
> works) and when received from the UA.
>
> The difference is in the branch= value as you predicted.
>
> How can I save the branch value on the INVITE and then change it on the
> CANCEL?
> I've tried all sorts of combinations for $branch but the vales are always
> NULL
>
>
> Regards
> Juri Nysschen
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
> Sent: Monday, February 14, 2011 10:16 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] FW: CANCELs with no transaction
>
> Juri,
>
> This was gnawing at me so I searched and found the "official" answer
> of what a CANCEL should look like.
> Go to
> http://www.ietf.org/rfc/rfc3261.txt
> and search for "9 Canceling a Request"
> From what I read it basically says that the CANCEL should be nearly
> identical to the INVITE it is canceling (without some headers that
> aren't useful in a cancel like Supported)
> It seems that the "branch=" , a "Magic Cookie", on the top most
> via line is a turbo lookup to finding the transaction. That matches
> the original INVITE and all is well. If it is not there then it has to
> compare all the pertinent headers between invite and cancel.
>
> So if you can't get them to send the proper stuff, you could modify
> the previous hack to store the branch=... that opensips put in the via
> header it added (probably can't see it when sending the invite but it
> is in the 1xx response to your invite.
> Actually I just found one better I think. Use the exec module to
> execute MI command t_uac_cancel
> (http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id294623)
> All it needs are callid and cseq num, which I hope at least they are
> the same as the INVITE, which you can use the CANCEL's parsed vars for
> the params. I you will probably need to send back a negative response
> to the bad CANCEL like 400 Bad Message (look up appropriate
> code/message). And I think the t_uac_cancel will also push into
> failure_route a 487 response, which if you don't choose anything else
> specifically, will be sent back to the originator.
> You will want to verify it with packet captures and see how it affects
> your cdrs.
>
> Let me know the outcome.
>
> Dave
>
> On Mon, Feb 14, 2011 at 11:05 AM, Dave Singer 
> wrote:
>> I was half expecting that kind of result.
>> I don't know what exactly opensips uses to match transactions. I
>> believe Call-id and CSeq headers being the same but I'm quite sure
>> there is more like maybe from tag? I don't know.
>>
>> Can anyone else chime in on what opensips is checking to match a
> transaction.
>>
>> Dave.
>>
>> On Sun, Feb 13, 2011 at 11:36 PM, Juri Nysschen 
> wrote:
>>> Hi Dave,
>>>
>>> Thanks this method is very successful in delivering the CANCEL
> downstream,
>>> over multiple hops, unfortunately the PSTN also doesn't see the
> transaction
>>> id and therefore the call keeps ringing.
>>> The key is finding the reason why the transaction id disappears or at
> least
>>> be able to put it back when delivering the CANCEL downstream.
>>>
>>> Regards
>>> Juri Nysschen
>>>
>>> -Original Message-
>>> From: users-boun...@lists.opensips.org
>>> [mailto:users-boun...@lists.opensips.org] On Behal

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

[Dave Singer]

Juri,

This was gnawing at me so I searched and found the "official" answer
of what a CANCEL should look like.
Go to
http://www.ietf.org/rfc/rfc3261.txt
and search for "9 Canceling a Request"
>From what I read it basically says that the CANCEL should be nearly
identical to the INVITE it is canceling (without some headers that
aren't useful in a cancel like Supported)
It seems that the "branch=" , a "Magic Cookie", on the top most
via line is a turbo lookup to finding the transaction. That matches
the original INVITE and all is well. If it is not there then it has to
compare all the pertinent headers between invite and cancel.

So if you can't get them to send the proper stuff, you could modify
the previous hack to store the branch=... that opensips put in the via
header it added (probably can't see it when sending the invite but it
is in the 1xx response to your invite.
Actually I just found one better I think. Use the exec module to
execute MI command t_uac_cancel
(http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id294623)
All it needs are callid and cseq num, which I hope at least they are
the same as the INVITE, which you can use the CANCEL's parsed vars for
the params. I you will probably need to send back a negative response
to the bad CANCEL like 400 Bad Message (look up appropriate
code/message). And I think the t_uac_cancel will also push into
failure_route a 487 response, which if you don't choose anything else
specifically, will be sent back to the originator.
You will want to verify it with packet captures and see how it affects
your cdrs.

Let me know the outcome.

Dave

On Mon, Feb 14, 2011 at 11:05 AM, Dave Singer  wrote:
> I was half expecting that kind of result.
> I don't know what exactly opensips uses to match transactions. I
> believe Call-id and CSeq headers being the same but I'm quite sure
> there is more like maybe from tag? I don't know.
>
> Can anyone else chime in on what opensips is checking to match a transaction.
>
> Dave.
>
> On Sun, Feb 13, 2011 at 11:36 PM, Juri Nysschen  
> wrote:
>> Hi Dave,
>>
>> Thanks this method is very successful in delivering the CANCEL downstream,
>> over multiple hops, unfortunately the PSTN also doesn't see the transaction
>> id and therefore the call keeps ringing.
>> The key is finding the reason why the transaction id disappears or at least
>> be able to put it back when delivering the CANCEL downstream.
>>
>> Regards
>> Juri Nysschen
>>
>> -Original Message-
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
>> Sent: Monday, February 14, 2011 9:03 AM
>> To: ty...@fonality.com; OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] FW: CANCELs with no transaction
>>
>> Juri,
>>
>> You say it only happens after a do_routing(). To be clear, you mean
>> that you used do_routing on the invite and not that you already tried
>> do_routing for the cancel earlier in the script. (I'm not sure it
>> would make any difference)
>>
>> The hack to store the destination in a variable is one you would want
>> to strongly avoid. But sometimes hacks are unavoidable and a stop gap
>> until you can really resolve the problem.
>> I believe $avp's are retained per transaction and if the
>> t_check_trans() fails then the $avp created in the reply would also
>> not be available.
>> $var also would not work most of the time since it is persistent per
>> process.  You would need to use core functions cache_store and
>> cache_fetch using either local_cache or memcached backend depending if
>> you need persistant between opensips reboot.
>> Example:
>>
>> route{
>>
>> .
>>
>>      if (is_method("CANCEL") ) {
>>
>>            route(5); # drop media proxy
>>
>>            if (t_check_trans()){ # this always fails after a do_routing()
>>
>>                  xlog("L_INFO","CANCEL
>> Transaction[$fd/$fu/$rd/$ru/$si/]\n");
>>
>>                  t_relay();
>>
>>                  exit;
>>
>>            } else {
>>
>>                 if ( cache_fetch("local", "tran_dest_$ci",
>> "$avp(s:next_hop)") ) {
>>                      $rd = $avp(s:next_hop);
>>                      t_relay();
>>                 }
>>            exit;
>>
>>      }
>>
>> }
>>
>>
>> on_reply[main] {
>>    cache_store("local", "tran_dest_$ci", "$si", 500);
>> }
>>
>> Da

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

[Dave Singer]

I was half expecting that kind of result.
I don't know what exactly opensips uses to match transactions. I
believe Call-id and CSeq headers being the same but I'm quite sure
there is more like maybe from tag? I don't know.

Can anyone else chime in on what opensips is checking to match a transaction.

Dave.

On Sun, Feb 13, 2011 at 11:36 PM, Juri Nysschen  wrote:
> Hi Dave,
>
> Thanks this method is very successful in delivering the CANCEL downstream,
> over multiple hops, unfortunately the PSTN also doesn't see the transaction
> id and therefore the call keeps ringing.
> The key is finding the reason why the transaction id disappears or at least
> be able to put it back when delivering the CANCEL downstream.
>
> Regards
> Juri Nysschen
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
> Sent: Monday, February 14, 2011 9:03 AM
> To: ty...@fonality.com; OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] FW: CANCELs with no transaction
>
> Juri,
>
> You say it only happens after a do_routing(). To be clear, you mean
> that you used do_routing on the invite and not that you already tried
> do_routing for the cancel earlier in the script. (I'm not sure it
> would make any difference)
>
> The hack to store the destination in a variable is one you would want
> to strongly avoid. But sometimes hacks are unavoidable and a stop gap
> until you can really resolve the problem.
> I believe $avp's are retained per transaction and if the
> t_check_trans() fails then the $avp created in the reply would also
> not be available.
> $var also would not work most of the time since it is persistent per
> process.  You would need to use core functions cache_store and
> cache_fetch using either local_cache or memcached backend depending if
> you need persistant between opensips reboot.
> Example:
>
> route{
>
> .
>
>      if (is_method("CANCEL") ) {
>
>            route(5); # drop media proxy
>
>            if (t_check_trans()){ # this always fails after a do_routing()
>
>                  xlog("L_INFO","CANCEL
> Transaction[$fd/$fu/$rd/$ru/$si/]\n");
>
>                  t_relay();
>
>                  exit;
>
>            } else {
>
>                 if ( cache_fetch("local", "tran_dest_$ci",
> "$avp(s:next_hop)") ) {
>                      $rd = $avp(s:next_hop);
>                      t_relay();
>                 }
>            exit;
>
>      }
>
> }
>
>
> on_reply[main] {
>    cache_store("local", "tran_dest_$ci", "$si", 500);
> }
>
> Dave
>
> On Sun, Feb 13, 2011 at 5:15 AM, Tyler Merritt  wrote:
>>
>> Why not use an $avp and grab the Call ID header on the inbound packet and
> then create some routing logic that checks the $avp against the return
> packet Call ID header to validate it's the same thing?  $avps can be made
> available onreply with a modparam though forgive me if it's a bit late at
> night and I don't have the link handy.
>> An avp can store more than a single value but they index in reverse order
> as written if I recall correctly.
>>
>> On Sat, Feb 12, 2011 at 5:05 AM, Russell Bierschbach
>  wrote:
>>>
>>> I have a similar problem, but not solution, my probably is actually
> occurring because the originating UA is ignoring a contact header that is
> sent back during a 183 progress message.  OpenSIPS uses information from
> that contact header to figure out where to relay the incoming message (BYE
> in my case, CANCEL in yours).  It seems like it would be possible for
> OpenSIPS to use a call-id or tag to determine where to relay the message
> though.
>>>
>>>
>>>
>>> Russell Bierschbach
>>>
>>> em: rbierschb...@telepointglobal.com, im: rbierschb...@hotmail.com
>>>
>>>
>>>
>>> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Juri Nysschen
>>> Sent: Friday, February 11, 2011 7:44 AM
>>> To: users@lists.opensips.org
>>> Subject: [OpenSIPS-Users] FW: CANCELs with no transaction
>>>
>>>
>>>
>>> Hi All,
>>>
>>>
>>>
>>> Need help with a nagging issue:
>>>
>>>
>>>
>>> UA->Opensips 1->Opensips 2->PSTN
>>>
>>>
>>>
>>> UA sends an invite on Opensips 1, and is routed via do_routing() to
> Opensips 2, Opensips 2 uses do_routing to get to the PST

Re: [OpenSIPS-Users] Weird behaviour

[Dave Singer]

Juri,

Thanks for the compliment!
Great addition yourself!
I hadn't looked at the premissions module before and it looks to be a
good resource for potentially many tasks.
Using it as you show would be much faster than trying to do db lookup
and cache_store method!
Only one thing, you are only showing to do it for OPTIONS, what about NOTIFY?

Dave,

On Sun, Feb 13, 2011 at 11:40 PM, Juri Nysschen  wrote:
> Hi All
>
> Filtering OPTIONs is brilliant, I've taken it one step further by also
> filtering against the opensips address table.
>
> So if the ip is listed in the table OR had registered beforehand there will
> be a reply on the OPTIONs message:
>
>    if (is_method("OPTIONS")){
>                xlog("L_INFO","OPTIONS [$fd/$fu/$rd/$ru/$si/]\n");
>                if (check_source_address("1")) ||
> (cache_fetch("local","ip_allowed_$si",$avp(i:60))){
>                        xlog("L_INFO","OPTIONS OK
> [$fd/$fu/$rd/$ru/$si/]\n");
>                sl_send_reply("200", "Got it");
>                exit;
>                };
>                drop;
>    };
>
>
> On a successful register:
>        cache_store("local","ip_allowed_$si","$si",1200);
>
>
> Regards
> Juri Nysschen
>
>
> -Original Message-
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Vasile
> Sent: Saturday, February 12, 2011 9:26 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Weird behaviour
>
> That's why I dropped the ideea of having numbered usernames.
>
> On Feb 11, 2011, at 10:45 PM, Dave Singer wrote:
>
>> Adrian,
>>
>> Probably want to only respond to registers that are to valid user
>> accounts, drop the rest, as they start scanning with like 100, 101,
>> ., 5000,  etc
>>
>> Dave
>>
>> On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile  wrote:
>>> Hi Dave,
>>>
>>> Yeah, you're right.. Basically allow only REGISTER requests from anywhere
> and
>>> the rest check the source ip.
>>> Great ideea.
>>>
>>> I will implement it as soon as possible.
>>>
>>> Thanks,
>>> Adrian Vasile
>>> y...@opennet.ro
>>>
>>>
>>> On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
>>>
>>>> Adrian,
>>>>
>>>> I was just thinking about the implementing no response for INVITE a
>>>> little more...
>>>> You would want to handle the response checking similar to the
>>>> register.  If not found in the cache where you check the location
>>>> table if there is a registered user at the source ip.
>>>> That way it can handle opensips reboots and other situations where the
>>>> cache is lost or unavailable. Like a memcached server fails.
>>>> Advantage to using external memcached vs local cache would be that
>>>> cache would not be cleared on opensips restart.
>>>>
>>>> Dave
>>>>
>>>> On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer
>  wrote:
>>>>> I've found that generally they start out with the sip NOTIFY or
>>>>> OPTIONS message. So recently I set in the script to drop them from
>>>>> sources I'm not expecting them from. Might not work so well for some
>>>>> situation like ATA's sending pings for nat keep alives. But for the
>>>>> nat to keep open, generally it doesn't need a response, just as long
>>>>> as they keep sending the packets. Some devices I've seen actually send
>>>>> essentially an empty packet to the sip port, just enough to keep the
>>>>> nat alive but opensips just discards it because it is empty.
>>>>> The one I do send a reply to is my network monitoring server. Kind of
>>>>> helpful to know when things stop responding. :-)
>>>>> If an ATA model need to actually get a reply you could on registration
>>>>> check the model listed in the sip agent header and use localcache or
>>>>> memcached to store the source IP as ok to respond to. See
>>>>> http://www.opensips.org/Resources/DocsCoreFcn16#toc98
>>>>> cache_store and cache_fetch
>>>>> at registration something like
>>>>>   save("location");
>>>>>   cache_store("local", "ping_ok_$si", "ok", 86000);
>>>>>  and at ping
>>>>>   if (

Re: [OpenSIPS-Users] Basic doubt of sip routing

[Dave Singer]

Toyima,
The client does NOT have to use NAPTR or SRV. It can use regular A or
CNAM records as well or even just an IP. (unless it is a half baked
client)
SRV and NAPTR are special DNS records that can be used tell the client
what services are available and where and how to connect to them. If
your starting out, don't worry about them. When you start to get into
failover and load balancing, then give them a look over.
If other service is providing them for opensips to use, nothing
special to do for opensips, it does that all automatically.

The most common use of opensips it to take a call request to a phone
number and decide where the next place to send that call request. A
super simplified two steps:
1 lookup if the called number belongs to one of your customers. If so
send it to them.
2 If no above, then lookup if the source of the call is one of your
customers. If so send the call through a carrier that can get it to
the owner of the number. (basically PSTN -- regular telephone
networks)

There are many ways that opensips can accomplish those tasks. As
Adrian said about choosing where to send it can be done, there are
multiple ways to check where the call came from and if want to send
the call on, like authentication via DB or radius, static source IP in
script, a module like dynamic_routing, or even a custom sql query.
It all depends on your needs.
If your just getting started, there are some good recommendations in
some fairly recent email threads for where to start using what
resources.

Dave

On Fri, Feb 11, 2011 at 5:16 AM, Toyima Dias  wrote:
> COOL Adrian...many thanks for your kindly answers...by the way, i've checked
> on the rfc that the client must use NAPTR and SRV to resolve domains!
>
> 2011/2/11 Adrian Georgescu 
>>
>> If your SIP device support dialing only phone numbers, you need a
>> translation mechanism, this you can implement in the SIP proxy. You can use
>> standard ENUM (http://www.faqs.org/rfc/rfc3764.txt),  local database
>> lookups, configuration logic to translate the number into a fully qualified
>> SIP address or many other methods, there are plenty of OpenSIPS modules that
>> do such phone number translation.
>> Adrian
>> On Feb 11, 2011, at 2:04 PM, Toyima Dias wrote:
>>
>> create...got it...that means that if i have a phone registered in proxy A,
>> and i want to call userB, A has no idea where B resides, at all...how does A
>> know the domain of B? he must put in the RURI of the invite
>> userB@domain_of_b, right? how does A knows the domain of B? does A must
>> press in the phone: usearB(a number)@DOMAIN_OF_B?
>>
>> sorry for many questions, keep reading the rfc...
>>
>> 2011/2/11 Adrian Georgescu 
>>>
>>> SIP routing works exactly like email. How did you know to email this
>>> list?
>>> Adrian
>>>
>>> On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote:
>>>
>>> Thanks Adrian...
>>> So...how does ALice now that bob is in the biloxi.com domain? per the rfc
>>> 3263 section 4 (client usage) the ua must use DNS to determine where to send
>>> a call...but i have a softphone righ now, and i'm trying to make a call like
>>> this:
>>>
>>> 234...@proxy2.com (inserted by me), not just puting the numbermaybe
>>> i'm saying something crazy, but i'm confuse!
>>>
>>> 2011/2/11 Adrian Georgescu 

 The SIP proxy lookups up the domain part, what appears after the @ sign
 before any parameters separated by ; if is an IP address like in your
 example you do not perform a DNS lookup you just send the packet there.
 In the request URI you must put the address of the remote end, not your
 own address. In your example user A calls user 264512380973 on the same
 Proxy and not a remote one.
 See the illustrated examples from:
 http://www.tech-invite.com/Ti-sip-ex3261.html
 Adrian
 On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote:

 Adrian, i'm checking the rfc...but even i have a question...when UA
 sends an INVITE to it's proxy to a phone for example (obviously not
 registered on the proxy), the proxy will check the RURI of this invite and
 it will se the following:

 user A sends the invite to its proxy : INVITE
 sip:264512380973@172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the
 IP of proxy A)
 Where does the DNS takes part? the domain is it's ip address...i'm quite
 confuse, any help would b e appreciated

 Thanks

 2011/2/11 Toyima Dias 
>
> Thanks Adrian...reading the RFC3263!
>
> Thanks!
>
> 2011/2/11 Adrian Georgescu 
>>
>> The proxy is using DNS to lookup the destination server.
>> Google for RFC 3263
>> Adrian
>>
>> On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote:
>>
>> Hello community,
>>
>> I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call
>> for a domain that he doesn't now? i mean...user A is registered in proxy 
>> AA,
>> if A wants to call to another user in another domain (not 

Re: [OpenSIPS-Users] FW: CANCELs with no transaction

[Dave Singer]

Juri,

You say it only happens after a do_routing(). To be clear, you mean
that you used do_routing on the invite and not that you already tried
do_routing for the cancel earlier in the script. (I'm not sure it
would make any difference)

The hack to store the destination in a variable is one you would want
to strongly avoid. But sometimes hacks are unavoidable and a stop gap
until you can really resolve the problem.
I believe $avp's are retained per transaction and if the
t_check_trans() fails then the $avp created in the reply would also
not be available.
$var also would not work most of the time since it is persistent per
process.  You would need to use core functions cache_store and
cache_fetch using either local_cache or memcached backend depending if
you need persistant between opensips reboot.
Example:

route{

.

  if (is_method("CANCEL") ) {

route(5); # drop media proxy

if (t_check_trans()){ # this always fails after a do_routing()

  xlog("L_INFO","CANCEL Transaction[$fd/$fu/$rd/$ru/$si/]\n");

  t_relay();

  exit;

} else {

 if ( cache_fetch("local", "tran_dest_$ci",
"$avp(s:next_hop)") ) {
  $rd = $avp(s:next_hop);
  t_relay();
 }
exit;

  }

}


on_reply[main] {
cache_store("local", "tran_dest_$ci", "$si", 500);
}

Dave

On Sun, Feb 13, 2011 at 5:15 AM, Tyler Merritt  wrote:
>
> Why not use an $avp and grab the Call ID header on the inbound packet and 
> then create some routing logic that checks the $avp against the return packet 
> Call ID header to validate it's the same thing?  $avps can be made available 
> onreply with a modparam though forgive me if it's a bit late at night and I 
> don't have the link handy.
> An avp can store more than a single value but they index in reverse order as 
> written if I recall correctly.
>
> On Sat, Feb 12, 2011 at 5:05 AM, Russell Bierschbach 
>  wrote:
>>
>> I have a similar problem, but not solution, my probably is actually 
>> occurring because the originating UA is ignoring a contact header that is 
>> sent back during a 183 progress message.  OpenSIPS uses information from 
>> that contact header to figure out where to relay the incoming message (BYE 
>> in my case, CANCEL in yours).  It seems like it would be possible for 
>> OpenSIPS to use a call-id or tag to determine where to relay the message 
>> though.
>>
>>
>>
>> Russell Bierschbach
>>
>> em: rbierschb...@telepointglobal.com, im: rbierschb...@hotmail.com
>>
>>
>>
>> From: users-boun...@lists.opensips.org 
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of Juri Nysschen
>> Sent: Friday, February 11, 2011 7:44 AM
>> To: users@lists.opensips.org
>> Subject: [OpenSIPS-Users] FW: CANCELs with no transaction
>>
>>
>>
>> Hi All,
>>
>>
>>
>> Need help with a nagging issue:
>>
>>
>>
>> UA->Opensips 1->Opensips 2->PSTN
>>
>>
>>
>> UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips 
>> 2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing.
>>
>>
>>
>> UA cancels call before answer, but now t_check_trans fails and the CANCEL is 
>> not passed onto the PSTN, with the result that the call rings forever and 
>> can only be terminated by the remote answering and dropping the call or 
>> through a timeout.
>>
>>
>>
>> The scripts on Opensips 1 & Opensips 2 is virtuall identical:
>>
>>
>>
>> How do I get the CANCEL to the PSTN ?
>>
>>
>>
>> route{
>>
>> .
>>
>>   if (is_method("CANCEL") ) {
>>
>>     route(5); # drop media proxy
>>
>>     if (t_check_trans()){ # this always fails after a do_routing()
>>
>>   xlog("L_INFO","CANCEL 
>> Transaction[$fd/$fu/$rd/$ru/$si/]\n");
>>
>>   t_relay();
>>
>>   exit;
>>
>>     };
>>
>>     exit;
>>
>>   }
>>
>> }
>>
>>
>>
>>
>>
>> route[4] {
>>
>>   xlog("L_INFO","Route4 [$fd/$fu/$rd/$ru/$si/]\n");
>>
>>
>>
>>   $avp(i:102)=1; # Default dr-group
>>
>>   route(10); # Do custom stuff
>>
>>   t_on_failure("4");
>>
>>   if (do_routing("$avp(i:102)")){
>>
>>     xlog("L_INFO","Route4 Route to Dyna Group: 
>> $avp(i:102)[$fd/$fu/$rd/$ru/$si/]\n");
>>
>>     t_newtran();
>>
>>     route(1);
>>
>>     exit;
>>
>>   };
>>
>>   xlog("L_INFO","Route4 No Route to Host[$fd/$fu/$rd/$ru/$si/]\n");
>>
>>   sl_reply_error();
>>
>>   exit;
>>
>> }
>>
>>
>>
>> Regards
>>
>> Juri Nysschen
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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Re: [OpenSIPS-Users] Weird behaviour

[Dave Singer]

Adrian,

Probably want to only respond to registers that are to valid user
accounts, drop the rest, as they start scanning with like 100, 101,
., 5000,  etc

Dave

On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile  wrote:
> Hi Dave,
>
> Yeah, you're right.. Basically allow only REGISTER requests from anywhere and
> the rest check the source ip.
> Great ideea.
>
> I will implement it as soon as possible.
>
> Thanks,
> Adrian Vasile
> y...@opennet.ro
>
>
> On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
>
>> Adrian,
>>
>> I was just thinking about the implementing no response for INVITE a
>> little more...
>> You would want to handle the response checking similar to the
>> register.  If not found in the cache where you check the location
>> table if there is a registered user at the source ip.
>> That way it can handle opensips reboots and other situations where the
>> cache is lost or unavailable. Like a memcached server fails.
>> Advantage to using external memcached vs local cache would be that
>> cache would not be cleared on opensips restart.
>>
>> Dave
>>
>> On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer  
>> wrote:
>>> I've found that generally they start out with the sip NOTIFY or
>>> OPTIONS message. So recently I set in the script to drop them from
>>> sources I'm not expecting them from. Might not work so well for some
>>> situation like ATA's sending pings for nat keep alives. But for the
>>> nat to keep open, generally it doesn't need a response, just as long
>>> as they keep sending the packets. Some devices I've seen actually send
>>> essentially an empty packet to the sip port, just enough to keep the
>>> nat alive but opensips just discards it because it is empty.
>>> The one I do send a reply to is my network monitoring server. Kind of
>>> helpful to know when things stop responding. :-)
>>> If an ATA model need to actually get a reply you could on registration
>>> check the model listed in the sip agent header and use localcache or
>>> memcached to store the source IP as ok to respond to. See
>>> http://www.opensips.org/Resources/DocsCoreFcn16#toc98
>>> cache_store and cache_fetch
>>> at registration something like
>>>   save("location");
>>>   cache_store("local", "ping_ok_$si", "ok", 86000);
>>>  and at ping
>>>   if ( $rm =~ "OPTIONS|NOTIFY" ) {
>>>     if( $si == "" || $cache_fetch("local",
>>> "pingok_$si", $avp(i:5)) {
>>>        sl_send_reply("200", "Ok");
>>>     }
>>>     drop;
>>>  }
>>>
>>> Might not need pike if they never start the brute force scan because
>>> they didn't get the initial reply.
>>> I just came up with this the other day so it is an unproved theory.
>>> The other day I left a packet capture running over night on the
>>> testing server and in the morning I saw all the failed register
>>> attempts. I looked back to the first packet from the registering
>>> source and found the first one was an OPTIONS packet and thus my
>>> theory.
>>>
>>> Could apply it to INVITE and other messages. For registrations if
>>> there wasn't a hit in the cache you would want to do a db lookup to
>>> see if the from user is one of yours. But generally that would only be
>>> for a first time registration since registrations usually happen every
>>> 30 min. (This is just brainstorming) ;-)
>>> Let me know if you implement some of it and what results you find.
>>>
>>> Dave
>>>
>>>
>>> On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile  wrote:
>>>> I know of these issues. And all client are either behind NAT either 
>>>> separate
>>>> voice vlans.
>>>> As for securing the proxy. What methods either than Pike combined with
>>>> fail2ban would you advise?
>>>>
>>>>
>>>> And I finally found the culprit. "Auth INVITE":
>>>> "When enabled, authorization is required for initial incoming INVITE
>>>> requests from the SIP proxy."
>>>> On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
>>>>
>>>> Adrian,
>>>>
>>>> There are lots of people out there with servers doing sip scans to see
>>>> if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
>>>> they will either tr

Re: [OpenSIPS-Users] Weird behaviour

[Dave Singer]

Adrian,

I was just thinking about the implementing no response for INVITE a
little more...
You would want to handle the response checking similar to the
register.  If not found in the cache where you check the location
table if there is a registered user at the source ip.
That way it can handle opensips reboots and other situations where the
cache is lost or unavailable. Like a memcached server fails.
Advantage to using external memcached vs local cache would be that
cache would not be cleared on opensips restart.

Dave

On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer  wrote:
> I've found that generally they start out with the sip NOTIFY or
> OPTIONS message. So recently I set in the script to drop them from
> sources I'm not expecting them from. Might not work so well for some
> situation like ATA's sending pings for nat keep alives. But for the
> nat to keep open, generally it doesn't need a response, just as long
> as they keep sending the packets. Some devices I've seen actually send
> essentially an empty packet to the sip port, just enough to keep the
> nat alive but opensips just discards it because it is empty.
> The one I do send a reply to is my network monitoring server. Kind of
> helpful to know when things stop responding. :-)
> If an ATA model need to actually get a reply you could on registration
> check the model listed in the sip agent header and use localcache or
> memcached to store the source IP as ok to respond to. See
> http://www.opensips.org/Resources/DocsCoreFcn16#toc98
> cache_store and cache_fetch
> at registration something like
>   save("location");
>   cache_store("local", "ping_ok_$si", "ok", 86000);
>  and at ping
>   if ( $rm =~ "OPTIONS|NOTIFY" ) {
>     if( $si == "" || $cache_fetch("local",
> "pingok_$si", $avp(i:5)) {
>        sl_send_reply("200", "Ok");
>     }
>     drop;
>  }
>
> Might not need pike if they never start the brute force scan because
> they didn't get the initial reply.
> I just came up with this the other day so it is an unproved theory.
> The other day I left a packet capture running over night on the
> testing server and in the morning I saw all the failed register
> attempts. I looked back to the first packet from the registering
> source and found the first one was an OPTIONS packet and thus my
> theory.
>
> Could apply it to INVITE and other messages. For registrations if
> there wasn't a hit in the cache you would want to do a db lookup to
> see if the from user is one of yours. But generally that would only be
> for a first time registration since registrations usually happen every
> 30 min. (This is just brainstorming) ;-)
> Let me know if you implement some of it and what results you find.
>
> Dave
>
>
> On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile  wrote:
>> I know of these issues. And all client are either behind NAT either separate
>> voice vlans.
>> As for securing the proxy. What methods either than Pike combined with
>> fail2ban would you advise?
>>
>>
>> And I finally found the culprit. "Auth INVITE":
>> "When enabled, authorization is required for initial incoming INVITE
>> requests from the SIP proxy."
>> On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
>>
>> Adrian,
>>
>> There are lots of people out there with servers doing sip scans to see
>> if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
>> they will either try to send register and/or invites for all sorts of
>> numbers trying to get a hit. Of course the invites are not actual
>> calls so if the sip scanner gets an ATA, the customer answers the
>> phone and there is no one there. Depending on the scanner it may keep
>> trying through it's whole list of common sip source accounts. Then it
>> can get interesting. The scanner would then mark the IP as a success
>> and the hacker can then start trying to send calls through it. Though
>> likely they would try a call to something like a Home Depot number and
>> when the customer answers they just say sorry wrong number and mark
>> the IP off their list. Customer is left alone till the next scanner
>> comes sniffing.
>> So ATA's many times have settings for not answering calls from places
>> that shouldn't be sending them calls. The options are usually
>> something like "calls ok: from register server, from proxy server,
>> call to registered user, auth call" or similar.
>> See what you can find in the docs for that model.
>>
>> Dave
>>
>> On Thu, Feb 10, 2011 at 5:07 AM, Adrian Vasile  w

Re: [OpenSIPS-Users] Weird behaviour

[Dave Singer]

I've found that generally they start out with the sip NOTIFY or
OPTIONS message. So recently I set in the script to drop them from
sources I'm not expecting them from. Might not work so well for some
situation like ATA's sending pings for nat keep alives. But for the
nat to keep open, generally it doesn't need a response, just as long
as they keep sending the packets. Some devices I've seen actually send
essentially an empty packet to the sip port, just enough to keep the
nat alive but opensips just discards it because it is empty.
The one I do send a reply to is my network monitoring server. Kind of
helpful to know when things stop responding. :-)
If an ATA model need to actually get a reply you could on registration
check the model listed in the sip agent header and use localcache or
memcached to store the source IP as ok to respond to. See
http://www.opensips.org/Resources/DocsCoreFcn16#toc98
cache_store and cache_fetch
at registration something like
   save("location");
   cache_store("local", "ping_ok_$si", "ok", 86000);
 and at ping
   if ( $rm =~ "OPTIONS|NOTIFY" ) {
 if( $si == "" || $cache_fetch("local",
"pingok_$si", $avp(i:5)) {
sl_send_reply("200", "Ok");
 }
 drop;
  }

Might not need pike if they never start the brute force scan because
they didn't get the initial reply.
I just came up with this the other day so it is an unproved theory.
The other day I left a packet capture running over night on the
testing server and in the morning I saw all the failed register
attempts. I looked back to the first packet from the registering
source and found the first one was an OPTIONS packet and thus my
theory.

Could apply it to INVITE and other messages. For registrations if
there wasn't a hit in the cache you would want to do a db lookup to
see if the from user is one of yours. But generally that would only be
for a first time registration since registrations usually happen every
30 min. (This is just brainstorming) ;-)
Let me know if you implement some of it and what results you find.

Dave


On Thu, Feb 10, 2011 at 10:28 AM, Adrian Vasile  wrote:
> I know of these issues. And all client are either behind NAT either separate
> voice vlans.
> As for securing the proxy. What methods either than Pike combined with
> fail2ban would you advise?
>
>
> And I finally found the culprit. "Auth INVITE":
> "When enabled, authorization is required for initial incoming INVITE
> requests from the SIP proxy."
> On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
>
> Adrian,
>
> There are lots of people out there with servers doing sip scans to see
> if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
> they will either try to send register and/or invites for all sorts of
> numbers trying to get a hit. Of course the invites are not actual
> calls so if the sip scanner gets an ATA, the customer answers the
> phone and there is no one there. Depending on the scanner it may keep
> trying through it's whole list of common sip source accounts. Then it
> can get interesting. The scanner would then mark the IP as a success
> and the hacker can then start trying to send calls through it. Though
> likely they would try a call to something like a Home Depot number and
> when the customer answers they just say sorry wrong number and mark
> the IP off their list. Customer is left alone till the next scanner
> comes sniffing.
> So ATA's many times have settings for not answering calls from places
> that shouldn't be sending them calls. The options are usually
> something like "calls ok: from register server, from proxy server,
> call to registered user, auth call" or similar.
> See what you can find in the docs for that model.
>
> Dave
>
> On Thu, Feb 10, 2011 at 5:07 AM, Adrian Vasile  wrote:
>
> Hi,
>
> I attached the trace.
>
>
> why does the cisco spa ask for authorization?
>
> Thanks,
>
> Adrian Vasile
>
> y...@opennet.ro
>
> On Feb 10, 2011, at 12:42 PM, Laszlo wrote:
>
> Hi Adrian,
>
> 2011/2/10 Adrian Vasile 
>
> Hello all,
>
> Maybe it has happened to you too.. I've got a couple of cisco spa504g
>
> everything is fine with them, registering, calling out, but there seems to
>
> be a problem with the "calling in feature"..
>
> When I try to call the spa's all they return is 403 Forbidden. Any ideas
>
> how I could remedy the situation?
>
>
> Try to capture one call with ngrep, and post here the output.
>
> Use ngrep like this: ngrep 'xxx' port 5060 -Wbyline -q -dany -t >
>
> mytrace.txt
>
> (where xxx is

Re: [OpenSIPS-Users] Weird behaviour

[Dave Singer]

Adrian,

There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send register and/or invites for all sorts of
numbers trying to get a hit. Of course the invites are not actual
calls so if the sip scanner gets an ATA, the customer answers the
phone and there is no one there. Depending on the scanner it may keep
trying through it's whole list of common sip source accounts. Then it
can get interesting. The scanner would then mark the IP as a success
and the hacker can then start trying to send calls through it. Though
likely they would try a call to something like a Home Depot number and
when the customer answers they just say sorry wrong number and mark
the IP off their list. Customer is left alone till the next scanner
comes sniffing.
So ATA's many times have settings for not answering calls from places
that shouldn't be sending them calls. The options are usually
something like "calls ok: from register server, from proxy server,
call to registered user, auth call" or similar.
See what you can find in the docs for that model.

Dave

On Thu, Feb 10, 2011 at 5:07 AM, Adrian Vasile  wrote:
> Hi,
> I attached the trace.
>
>
> why does the cisco spa ask for authorization?
> Thanks,
> Adrian Vasile
> y...@opennet.ro
>
> On Feb 10, 2011, at 12:42 PM, Laszlo wrote:
>
> Hi Adrian,
>
> 2011/2/10 Adrian Vasile 
>>
>> Hello all,
>>
>> Maybe it has happened to you too.. I've got a couple of cisco spa504g
>> everything is fine with them, registering, calling out, but there seems to
>> be a problem with the "calling in feature"..
>>
>> When I try to call the spa's all they return is 403 Forbidden. Any ideas
>> how I could remedy the situation?
>>
>
> Try to capture one call with ngrep, and post here the output.
> Use ngrep like this: ngrep 'xxx' port 5060 -Wbyline -q -dany -t >
> mytrace.txt
>
> (where xxx is the number/extension what you going to trace)
>
>
>
>>
>> Thanks,
>> Adrian Vasile
>> y...@opennet.ro
>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

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Re: [OpenSIPS-Users] b2b top-hiding and SDP origin line

[Dave Singer]

Jeff,

I checked nathelper code in opensips 1.6.4 and looks like adding the
"oldip" header to the SDP is not there anymore.
So you shouldn't need to modify the source like I had to previously to
accomplish complete topology hiding.
Also it doesn't really matter if you use rtp_proxy or media proxy
combined with fix_nated_sdp(8,"").
Which ever fits better for you.

Ovidiu,
According to change logs looks like you fixed that. Thanks

Dave

On Tue, Feb 8, 2011 at 6:20 PM, Jeff Pyle  wrote:
> "Now you now, and knowing is half the battle."
>                               -- G.I. Joe
>
>
> On 2/8/11 5:36 PM, "Ovidiu Sas"  wrote:
>
>>yup
>>
>>On Tue, Feb 8, 2011 at 5:15 PM, Jeff Pyle  wrote:
>>> Got it.
>>>
>>> Am I to believe, then, that only the initial "initialized" UAC INVITE
>>>hits
>>> the local route?  Everything after that for a dialog is visible in the
>>>b2b
>>> ones only, and therefore untouchable?
>>>
>>>
>>> - Jeff
>
>
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Re: [OpenSIPS-Users] every user is registered

[Dave Singer]

mato848,

The default/sample config has a section for registration. you need to
basically use that. just copy it to your script. Look for
www_authorize.
Registrations are stored with the usrloc module and auth is done by
auth and auth_db modules so they need to be configured as well like
the sample config.
Look at the docs for how you want to deal with multiple registrations:
   Allow them and what is an update or a different device (some
devices don't properly identify themselves according to SIP standards
so opensips may think it is a different device from a previous
registration and you end up with two registrations for the same device
and then opensips sends out two invites to the device)
   Call all simultaneously or one after another.

Dave

On Wed, Feb 9, 2011 at 10:40 AM, mato...@gmail.com  wrote:
> Hi,
> First, sorry for my english :)
> I have a problem with the user registration. I added some users to the
> database and tried to register with Twinkle and it worked. But when i tried
> to register other users (that were not in the database), they were
> registered too, they could make calls, IMs ... . Also each random user can
> use the server. Can anybody help me with that ?
>
>
>
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Re: [OpenSIPS-Users] Control Panel Dialog config for multiple servers

[Dave Singer]

Cinthia,

I'm fairly new to OCP myself, just got it running about a month ago.
Since no one with more experience responded, I'll help if I can.
Sounds like you have it all setup and working as designed.
Looking at my system that I have connected to two servers, it looks
like the dialog module section of OCP just is not yet developed to use
the server selection drop down like the MI section.
I'm not using dialog on the servers that I'm connecting it to so I can
see what the report looks like. Have you using the dialog MI commands
like dlg_list in the MI page? Is the output acceptable for your
purposes?

You could use the MI page as an example and modify the php scripts in
OCP to implement the server drop down functionality.
I would be willing to do it for some compensation and submit it to the
project on completion.
I modifying OCP a fair bit so I could use ssh with a key file to
connect to a remote host and run the fifo commands through the
opensipsctl script because it took less time to get that to work than
getting xml_rpc to compile and work in opensips.
It is in a state of it is working great for me. :-) I'm planning on
submitting my enhancements to the project after making it work more
generically (using boxes.global.inc.php fully rather than hard coding
some of my settings) and its worked for a while with me poking and
prodding it.

Hope this helps,
Dave

On Tue, Feb 8, 2011 at 6:35 PM, Daniel Goepp  wrote:
> I have not seen a response to this question yet, and it is of particular
> interest to me.  I don't suppose anyone has any comments on getting OCP to
> query multiple servers.  Thoughts?
>
> Much appreciated.
>
> -dg
>
>
> On Fri, Jan 28, 2011 at 4:46 PM, Cinthia Leung  wrote:
>>
>> Hello,
>>
>> I have several OpenSIPS servers running and I'd like to have OCP installed
>> on a separate server to keep track of all of them.  At this time I'm stuck
>> at configuring the dialog tool.
>>
>> It looks like it can send MI commands to multiple servers to get a list of
>> active dialogs, is that right?
Yes.
>> If yes, I can use some help here. I am unable to grab dialog status from
>> all servers, only one at a time.
You are able now to get the dialog status from each server one at a
time or you believe it should.
Your question
>>
>> I think I have boxes.global.inc.php configured right, since I'm able to
>> see all my servers in the drop down list of the System -> MI Commands tool
>> page
>>
>> I don't know if I have config/tools/system/dialog/local.inc.php set
>> correctly.  I have these lines in the file
>>  $box[0]['mi']['conn']="server1_ip:8080";
>>  $box[1]['mi']['conn']="server2_ip:8080";
>> ...
>>
>> Please help.  Thanks.
>>
>>
>> Cinthia
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Re: [OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

[Dave Singer]

So weird.
Did you specify the -f  on the opensips cmd line?
That is the only thing left that I can think of as the problem, it is
using a default config that is somewhere else.

try
  updatedb
  locate opensips.cfg
Most likely the default location is /usr/etc/opensips/opensips.cfg or
/etc/opensips/opensips.cfg, depending on compile time options.

Failing that I would scrub out opensips installed pieces and
recompile/reinstall with a different --prefix=

Dave

On Tue, Feb 8, 2011 at 11:51 AM, Chris Stone  wrote:
> Dave,
>
> On Tue, Feb 8, 2011 at 12:02 AM, Dave Singer  
> wrote:
>> Don't know what tools you are familiar with so here are some
>> suggestions for what they're worth.
>
> Appreciate the input!
>
> Am familiar with all - but included output below - always happy to
> have another set of eyes on things  ;-)
>
>> what is listening on port 5060?
>>   netstat -lnp | grep 5060
>
> tcp        0      0 67.212.153.178:5060         0.0.0.0:*
>     LISTEN      25098/opensips
> tcp        0      0 127.0.0.1:5060              0.0.0.0:*
>     LISTEN      25098/opensips
> udp        0      0 67.212.153.178:5060         0.0.0.0:*
>                 25098/opensips
> udp        0      0 127.0.0.1:5060              0.0.0.0:*
>                 25098/opensips
>
> and as seen below, that's the only opensips pid on the system...
>
>> is opensips actually running? what was the command line used?
>>   ps aux --forest | grep opensips
>
> root     25098  0.0  0.0  72884   992 ?        S    11:40   0:00
> /usr/sbin/opensips
> root     25100  0.0  0.0  72884   468 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25101  0.0  0.0  72884   468 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25102  0.0  0.0  72884   468 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25103  0.0  0.0  72884   468 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25104  0.0  0.0  72884   468 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25105  0.0  0.0  72884   472 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25106  0.0  0.0  72884   472 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25107  0.0  0.0  72884   472 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25108  0.0  0.0  72884   472 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25109  0.0  0.0  72884   472 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25110  0.0  0.0  72884   592 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25111  0.0  0.0  72884   600 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25112  0.0  0.0  72884   708 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25113  0.0  0.0  72884   708 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25114  0.0  0.0  72884   708 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25115  0.0  0.0  72884   708 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25116  0.0  0.0  72884   708 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
> root     25117  0.0  0.0  72884   704 ?        S    11:40   0:00  \_
> /usr/sbin/opensips
>
>> Does sip still pass when opensips is not running?
>
> Nope - inbound calls hang and then get a fast busy. Start opensips and
> they ring through.
>
>> if mi_fifo loaded, what is the output of > path>/sbin/opensipsctl fifo ps
>> does it show that it is listening on the interfaces/ports that are
>> handeling the packets you are capturing?
>
> Wasn't loaded. Added it and retested call and yes, it shows listening
> on the correct interfaces and ports - all interfaces and tcp/udp port
> 5060:
>
> Process::  ID=0 PID=25256 Type=attendant
> Process::  ID=1 PID=25257 Type=SIP receiver udp:127.0.0.1:5060
> Process::  ID=2 PID=25259 Type=SIP receiver udp:127.0.0.1:5060
> Process::  ID=3 PID=25260 Type=SIP receiver udp:127.0.0.1:5060
> Process::  ID=4 PID=25261 Type=SIP receiver udp:127.0.0.1:5060
> Process::  ID=5 PID=25262 Type=SIP receiver udp:127.0.0.1:5060
> Process::  ID=6 PID=25263 Type=SIP receiver udp:67.212.153.178:5060
> Process::  ID=7 PID=25264 Type=SIP receiver udp:67.212.153.178:5060
> Process::  ID=8 PID=25265 Type=SIP receiver udp:67.212.153.178:5060
> Process::  ID=9 PID=25266 Type=SIP receiver udp:67.212.153.178:5060
> Process::  ID=10 PID=25267 Type=SIP receiver udp:67.212.153.178:5060
> Process::  ID=11 PID=25268 Type=time_keeper
> Process::  ID=12 PID=25269 Type=timer
> Process::  ID=13 PID=25270 Type=MI FIFO
> Process::  ID=14 PID=25271 Type=TCP receiver
> Process::  ID=15 PID=25272 Type=TCP receiver
> Process::  ID=16

Re: [OpenSIPS-Users] b2b top-hiding and SDP origin line

[Dave Singer]

Jeff,

I ran into this too. I was not using opensips to do the SIP top hiding
but was for the SDP using media_proxy. Media proxy doesn't touch "o"
unfortunately.
So I used fix_nated_sdp :
  fix_nated_sdp("8","$Ri"); #the recieved on IP (my IP)
That fixed the "o" header but it created a "oldip" or "origip" or
something like that in the sdp.
So I ended up modifying the source to not add the extra sdp header and
it worked good.
Not sure this will work for your situation but worth mentioning.
It would be nice if media_proxy and nat_helper had params for changing
the "o" header at the same time as the "c".
I looked at doing that in source but it was a little beyond my confort zone. :-)

Dave


On Tue, Feb 8, 2011 at 11:40 AM, Jeff Pyle  wrote:
> Hello,
> Since the top-hiding scenario doesn't touch the SDP, it seems some
> extracurricular textops may be required to fully hide the topology of the
> network.  I've been trying various subst_body() functions on my b2b Opensips
> instance, each failing in a new and wonderful way.
>
> route[b2b_request] {
>         if (search("Content-Type: application/sdp")) {
>                 xlog("L_INFO", "Attempting to rewrite origin line in
> b2b_request\n");
>                 subst_body('/^o=.*/o=request 12345 67890 IN IP4 8.4.6.2/');
>         }
> }
> route[b2b_reply] {
>         if (search("Content-Type: application/sdp")) {
>                 xlog("L_INFO", "Attempting to rewrite origin line in
> b2b_reply\n");
>                 subst_body('/^o=.*/o=reply 12345 67890 IN IP4 8.4.6.2/');
>         }
> }
> local_route {
>         if (search("Content-Type: application/sdp")) {
>                 xlog("L_INFO", "Attempting to rewrite origin line in
> local_route\n");
>                 subst_body('/^o=.*/o=local 12345 67890 IN IP4 8.4.6.2/');
>         }
> }
>
> With the above config, the initial INVITE b2b'd through
> with b2b_init_request("top hiding") does have it's origin line operated
> upon.  After that, I see two xlogs hit from within b2b_reply on the 183 and
> 200 coming back through, but the subst_body commands have no effect.
> No, I'm not really going to rewrite my origin lines to a static value like
> this; this is just for testing.  And the test is not going well.
> How does one affect the origin line in a b2b environment for true topology
> hiding?  Or, is that the key, that it must be done outside of a b2b
> environment?
>
> - Jeff
>
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Re: [OpenSIPS-Users] Notify Message

[Dave Singer]

Kyle,

use t_replicate("sip:");
Look at the tm module docs for a more complete picture.
Then modify the message. One branch will go out with the message the
way it was before the t_replicate and one will have the changes after
the t_replicate.
If you need more replicated destinations, use append_branch same as
t_replicate with the final dest using the t_replicate cmd. For this
you might also need to move the message modifications to a branch
route because somethings can only be changed once like
uac_replace_from.

Dave

On Tue, Feb 8, 2011 at 9:45 AM, Kyle Haefner  wrote:
> Hi Users!
>
> I have a *simple* scenario that I just can't seem to
> accomplish...famous last words right?  Ok here goes...
>
>
> I receive a NOTIFY message that I want to forward to two destinations,
> one destination I want to forward the message and modify the body, the
> other destination I just want to forward the message.  Here is what I
> have so far it works for the modified message:
>
>
> if (method=="NOTIFY"){
>
> xlog("\n\nSIP NOTIFY
> *\n\n
>
> if(subst_body('/(Message-Account:.*)(sip:)([125][0-9][0-9][0-9][0-9])(@.*)$/\1\297049\3\4/')){
>                                                sl_send_reply("200", "OK");
>                                                t_relay("tcp:192.168.0.3");
>                                        };
>                                       ");
>                                        exit;
>
>                        }
>
> My initial attempt, doesn't work 192.168.0.3 always gives me a 500 error:
>
> if (method=="NOTIFY"){
>
> xlog("\n\nFORWARD ORIGINAL SIP NOTIFY
> *\n\n
>                                      sl_send_reply("200", "OK");
>                                      t_relay("tcp:192.168.0.2");
>
>
>
> xlog("\n\nFORWARD MODIFIED SIP NOTIFY
> *\n\n
>
> if(subst_body('/(Message-Account:.*)(sip:)([125][0-9][0-9][0-9][0-9])(@.*)$/\1\297049\3\4/')){
>                                                sl_send_reply("200", "OK");
>                                                t_relay("tcp:192.168.0.3");
>                                        };
>                                       ");
>                                        exit;
>
>                        }
>
>
>
> It seems that I should use append_branch for this, but then how do I
> modify the body of only one of the branches, and what is the best way
> to call t_relay?
> I'd appreciate it if someone could point me in the right direction!
>
> Regards,
>
> Kyle
>
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Re: [OpenSIPS-Users] How to stop script execution?

[Dave Singer]

Anton,

My bet is that since the first one was dropped, a transaction was
never created and thus the transaction cleanup doesn't happen either.
If you want it to it do the automatic cleanup it might work to create
a transaction prior to the reply, then use t_reply and finally drop.

Dave

On Tue, Feb 8, 2011 at 3:46 AM, Anton Zagorskiy
 wrote:
> Log inspections shows that while I processing first INVITE an original UAc
> sends INVITE again, but OpenSIPS doesn't recognize it as already received
> and start processing it again (makes a new transaction).
>
> Why that can happens?
>
>
>>
>> Hi.
>>
>> I have 2 nested routes:
>>
>> route -> route[invite] -> route[check]
>> In the route[check] I need to stop the execution of the script. I
>> wrote:
>>
>> route[check]
>> {
>>       sl_send_reply("401", "Forbidden");
>>       drop;
>> }
>>
>> But this isn't working.
>> OpenSIPS replies "401: Forbidden" but continues the execution of the
>> script.
>>
>> Is this normal?
>>
>>
>>
>>
>>
>>
>> WBR, Anton Zagorskiy
>> VoIP Developer, Oyster Telecom
>> Phone.: +7 812 601-0666
>> Fax: +7 812 601-0593
>> a.zagors...@oyster-telecom.ru
>> www.oyster-telecom.ru
>>
>>
>>
>>
>>
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Re: [OpenSIPS-Users] How to modify the 100 Trying packet

[Dave Singer]

I'm just trying to be helpful where I can, appreciative of the help
I've received and the great free software. My biggest worry is
misleading someone now or later in a list search. :-)

Just came across in another thread about the server header that you
can set it. Not sure of all the ramifications, I haven't tried it
myself.
Check out http://www.opensips.org/Resources/DocsCoreFcn16#toc66
You may want to try setting it to "" and see what happens. Though that
will affect any place it would use the Server header. Though I think
it is only a FYI header that is not required.

Seems like a very pick carrier. :-/  Oh the hoops they sometimes put
you through.

Dave

On Mon, Feb 7, 2011 at 8:14 PM, Tyler Merritt  wrote:
> Dave,
> Thanks for your reply.  I see you reply a lot and I appreciate you taking
> the time to help us.
> I read the page you linked - and I've stumbled across that before, but it's
> not what I want to do sadly :(
> The carrier basically wants me to strip the "Server" header from the 100
> message.  That's it - if this message went through the routing logic, I
> could just "remove_hf("Server");" but alas...
> I've told them this is a no-go for now.  Seems that we would have to modify
> the OpenSIPs core to make it happen.  Not worth the time or effort for this
> reason.
>
>
> On Tue, Feb 8, 2011 at 3:10 AM, Dave Singer 
> wrote:
>>
>> The 100 Trying is sent out to the originating server/user
>> automatically when you send the invite off to the next destination
>> with t_relay and exit in the main routing script.
>> The 100 Trying you receive in the on_reply is just to let you know
>> that the next hop is not dead and it is working on it. Your server
>> only uses it for it's timeout for the call to that server.
>> See the optional flags in the t_relay docs:
>> http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id271564
>> That should get you going.
>> FYI if (is_method("TRYING")) would always be false since the method is
>> INVITE and the status message would be Trying or "Giving a try" or
>> something else. Never count on the status message, just the code.
>> Different servers send back different messages. Unless you are
>> factoring that in.
>> I think you actually might be looking for fix_nated_contact. Check
>> that out. Depends on what you are trying to accomplish.
>>
>> Dave
>>
>> On Mon, Feb 7, 2011 at 3:48 AM, Tyler Merritt  wrote:
>> > This started working - but the problem is that it's matching both sides
>> > of
>> > the conversation.  I'm searching for a concrete method to say "only if
>> > you're going to send this packet to the upstream provider" but the
>> > closest I
>> > found was "is_destination" which has to be called from loose_route, and
>> > unfortunately it looks like we don't have any HF setup for this
>> > particular
>> > situation, so loose_route always returns 0 and exits (confirmed in the
>> > logs).
>> > The provider won't let me add stuff - maybe I can just add stuff one-way
>> > then rip it off on the return?
>> > if (is_method("INVITE") && status=="200" &&
>> > $(ct.fields(uri){s.substr,4,12})
>> > !~ '^8150*') {
>> >                                 xlog("L_INFO", "Modifying the outbound
>> > Contact header to Carrier...  $(ct.fields(uri){s.substr,5,12})");
>> >
>> >  subst('/^Contact:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/Contact:
>> > /ig');
>> > }
>> > Any tips?
>> > On Mon, Feb 7, 2011 at 11:19 AM, Tyler Merritt 
>> > wrote:
>> >>
>> >> I have used "if (status=="100")" and "if (is_method("TRYING"))" and
>> >> placed
>> >> these in a variety of places, at the top of route[0], in route[1], in
>> >> onreply_route (where it actually works), but nowhere can I get the
>> >> OUTGOING
>> >> 100 TRYING packet to change...
>> >> I can't figure this out - there isn't very good documentation
>> >> surrounding
>> >> this particular packet, and if I were to guess, I would say that the
>> >> method
>> >> is hard-coded somewhere in the routing module.
>> >> Any pointers on where this packet might be modified before sending it
>> >> to
>> >> the upstream provider would be welcome.
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>> >
>
>

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Re: [OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

[Dave Singer]

On Mon, Feb 7, 2011 at 7:05 PM, Ovidiu Sas  wrote:
> On Mon, Feb 7, 2011 at 9:14 PM, Chris Stone  wrote:
>> Sorry all for the last message - too quick on the Send button.
>>
>> On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink  
>> wrote:
>>> Hi Chris,
>>>
>>> That config should't touch the Contact header, and yet that's also been
>>> modified:
>>>
>>> In:  Contact:>> Out: Contact:>>
>>> Are you sure nothing else is touching the message?
>>
>> Yes, absolutely. The packets were captured on the Opensips server - in
>> from upstream provider and then the next packet relaying the invite to
>> backend Asterisk server. The upstream provider is, of course, on a
>> remote network. The Asterisk server is on the same LAN - only a switch
>> separating the 2 servers. The only application on the server that
>> touched the packets would be Opensips.
>>
>> So I'm not nuts - something very weird is going on here eh?
>
> Yes.  For sure you are using a different config.
> Are you running multiple servers on that box?
> Or are you having some virtual machines there?
>
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>
Don't know what tools you are familiar with so here are some
suggestions for what they're worth.
what is listening on port 5060?
   netstat -lnp | grep 5060
is opensips actually running? what was the command line used?
   ps aux --forest | grep opensips
Does sip still pass when opensips is not running?
if mi_fifo loaded, what is the output of /sbin/opensipsctl fifo ps
does it show that it is listening on the interfaces/ports that are
handeling the packets you are capturing?
If something else is listening on port 5060 opensips should be failing
to start with the provided config.
try running single mode with the following config:

#---
debug=9  # debug level (cmd line: -dd)
fork=no
log_stderror=yes  # (cmd line: -E)

children=1
check_via=no  # (cmd. line: -v)
dns=off   # (cmd. line: -r)
rev_dns=off   # (cmd. line: -R)
port=5060

# for more info: opensips -h

# -- module loading --
mpath="/usr/lib64/opensips/modules"
#loadmodule "xlog.so" # un comment if using opensips before 1.6.4
# - setting module-specific parameters ---


route{
   xlog("\n\n\ntesttest \n\n\n");
   forward("67.212.153.179");
   exit;
}

and run it with:
 /usr/sbin/opensips -f  2>&1 | tee test.log
(path to opensips executable based on your mpath in the config.

you should get a boat load of debug go past your screen when it starts
and for any message that goes through opensips.
To stop it just do CTRL+C
The debug went in to test.log as well as to the screen so you can look
in it and search for the "testtest" to see if it handled the
message. Or details of why it failed to start.

Happy hunting.
Dave

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Re: [OpenSIPS-Users] trying to rewriteuri to the value of a variable

[Dave Singer]

Last week I ran into a similar thing with force_socket(). I was trying
to setup to force it out the second IP on the server and have it
generic for two load balanced servers. I did if ( $Ri == "" ) { force_socket(":5060"); } else {
force_socket("server 2's IP>:5060"); }
Interesting that config file check didn't catch the "error" but when I
started it up for "real" it displayed an error and stopped running.
It is a good thing I had my 2nd level config check implemented on
there where I run it listening on a local interface and dummy port. It
failed that test so it didn't actually restart opensips. (FYI: I
poseted my init.d script a while back)
It took me a while to realize $fs was there and made things sooo
much better.

We need to document for the functions like rewriteuri where there are
alternative variables like $ru, $du, $fs that that the variables are
available and the difference between the two. Maybe even depricate the
less feature-full functions unless there is an advantage like
performance.

Dave

On Sun, Feb 6, 2011 at 7:01 PM, Duane Larson  wrote:
> I don't know that its mentioned in the documentation, but I ran across the
> same issue and remembered finding something about it by searching nabble's
> opensips part.
>
> Its one of those things you chalk up to experience.
>
> Sent from Droid
>
> On Feb 6, 2011 8:54 PM, "Nigel Daniels" 
> wrote:
>> that seems to be working, thanks alot ! clearly I need to re-read the docs
>>
>> On Sun, Feb 6, 2011 at 6:48 PM, Dave Singer
>> wrote:
>>
>>> Isn't
>>> $var(my_uri) = "5...@example.com";
>>> $ru = "sip:" + $var(my_ruri);
>>> the same as
>>> rewriteuri("sip:5...@example.com");
>>> just that using $ru you can use it just like assigning to other vars
>>> like you are doing.
>>>
>>>
>>> On Sun, Feb 6, 2011 at 6:36 PM, Duane Larson 
>>> wrote:
>>> > Avpops
>>> >
>>> > Sent from Droid
>>> >
>>> > On Feb 6, 2011 8:34 PM, "Nigel Daniels" <
>>> nigel.dani...@connectandsell.com>
>>> > wrote:
>>> >> the problem is that rewriteuri() does not support variables is there
>>> >> any
>>> >> alternative to rewriteuri() for changing the uri
>>> >>
>>> >> On Sun, Feb 6, 2011 at 6:26 PM, Tyler Merritt 
>>> wrote:
>>> >>
>>> >>> I do this as well:
>>> >>>
>>> >>> $var(origFromNumber) = $(re{uri.user});
>>> >>> $var(finalFromURI) = "sip:81" + $(var(origFromNumber){s.substr,1,0})
>>> >>> +
>>> >>> "@"
>>> >>> + $fd;
>>> >>>
>>> >>> This works fine for me. Takes what is essentially a DID and formats
>>> >>> it
>>> to
>>> >>> a SIP PEER string as per the carrier.
>>> >>>
>>> >>> You say that var(z) prints correctly in the logs, but according to
>>> >>> the
>>> >>> parser it's not a legit URI. It may just be that your upstream routes
>>> >>> don't
>>> >>> recognize this format.
>>> >>>
>>> >>> We use the usr_loc and db_aliases in order to match the URI up with a
>>> >>> destination - have you ensured that there is a valid destination for
>>> this
>>> >>> SIP URI?
>>> >>>
>>> >>> ___
>>> >>> Users mailing list
>>> >>> Users@lists.opensips.org
>>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >>>
>>> >>>
>>> >>
>>> >>
>>> >> --
>>> >> Nigel Daniels
>>> >> Network & Systems Administrator
>>> >> ConnectAndSell inc.
>>> >> (650)-533-2542 <+16505332542>
>>> >
>>> > ___
>>> > Users mailing list
>>> > Users@lists.opensips.org
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>> >
>>> >
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>>
>> --
>> Nigel Daniels
>> Network & Systems Administrator
>> ConnectAndSell inc.
>> (650)-533-2542
>
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>
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Re: [OpenSIPS-Users] How to test if a message is from myself

[Dave Singer]

Thanks Stefano,
Though, X-src-ip would be a header added by a trusted back 2 back
server (which I would fist verify before trusting the header). So it
wouldn't ever be the same as $si.

Thanks Bogdan,
I thought of a workaround. :-) (a little messy though)
if ( $avp(s:dr_gw_attrs) =~ "origin_ips=" ) {
$var(origin_ips) =
$(avp(s:dr_gw_attrs){param.value,origin_ips});
$var(i) = 0;
$var(found) = 0;
while ( $(var(origin_ips){s.select,$var(i),:}) != null
&& ! $var(found) ) {
$var(potential_fs_ip) =
$(var(origin_ips){s.select,$var(i),|});
$var(tmp_rd) = $rd;
$rd = $var(potential_fs_ip);
if ( uri == myself ) {
$fs =  $var(potential_fs_ip) + ":5060";
$var(found) = 1;
} else
$var(i) = $var(i) + 1;
}
if ( $var(i) ) {
$rd = $var(tmp_rd);
if ( ! $var(found) )
return(-1);
}
}

On Thu, Feb 3, 2011 at 4:32 AM, Stefano Pisani
 wrote:
> Hi Dave
> you could try
>
> if ($si == $hdr(X-src-ip)){...}
>
>
> Il 03/02/2011 12:59, Bogdan-Andrei Iancu ha scritto:
>>
>> Hi Dave,
>>
>> Unfortunately does not work with variables.
>>
>> Regards,
>> Bogdan
>>
>> Dave Singer wrote:
>>>
>>> Wow I missed that one. Thanks.
>>> Does that work for PVs so I can test other IPs like one from another
>>> header, say "X-src-ip:192.168.0.5". Last I tried I couldn't get it to
>>> work. Not sure if that was 1.6.2 or 1.6.3. I'm using 1.6.4 now. :)
>>>
>>> Thanks Again
>>> Dave
>>>
>>> On Wed, Feb 2, 2011 at 4:37 AM, Bogdan-Andrei Iancu 
>>> wrote:
>>>>
>>>> Hi Dave,
>>>>
>>>> do :  if (src_ip==myself) {}
>>>>
>>>> Regards,
>>>> Bogdan
>>>>
>>>> Dave Singer wrote:
>>>>>
>>>>> Is there any way to check if the source IP/port is one that opensips
>>>>> is listening on or one ? something like if ("sip:$si:$sp" == myself) {
>>>>> ...bla; bla;}
>>>>>
>>>>> Thanks
>>>>> Dave
>>>>>
>>>>> ___
>>>>> Users mailing list
>>>>> Users@lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>> --
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Event - expo, conf, social, bootcamp
>>>> 2 - 4 February 2011, ITExpo, Miami,  USA
>>>> OpenSIPS solutions and "know-how"
>>>>
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>
> ___
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>

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Re: [OpenSIPS-Users] How to modify the 100 Trying packet

[Dave Singer]

The 100 Trying is sent out to the originating server/user
automatically when you send the invite off to the next destination
with t_relay and exit in the main routing script.
The 100 Trying you receive in the on_reply is just to let you know
that the next hop is not dead and it is working on it. Your server
only uses it for it's timeout for the call to that server.
See the optional flags in the t_relay docs:
http://www.opensips.org/html/docs/modules/1.4.x/tm.html#id271564
That should get you going.
FYI if (is_method("TRYING")) would always be false since the method is
INVITE and the status message would be Trying or "Giving a try" or
something else. Never count on the status message, just the code.
Different servers send back different messages. Unless you are
factoring that in.
I think you actually might be looking for fix_nated_contact. Check
that out. Depends on what you are trying to accomplish.

Dave

On Mon, Feb 7, 2011 at 3:48 AM, Tyler Merritt  wrote:
> This started working - but the problem is that it's matching both sides of
> the conversation.  I'm searching for a concrete method to say "only if
> you're going to send this packet to the upstream provider" but the closest I
> found was "is_destination" which has to be called from loose_route, and
> unfortunately it looks like we don't have any HF setup for this particular
> situation, so loose_route always returns 0 and exits (confirmed in the
> logs).
> The provider won't let me add stuff - maybe I can just add stuff one-way
> then rip it off on the return?
> if (is_method("INVITE") && status=="200" && $(ct.fields(uri){s.substr,4,12})
> !~ '^8150*') {
>                                 xlog("L_INFO", "Modifying the outbound
> Contact header to Carrier...  $(ct.fields(uri){s.substr,5,12})");
>
>  subst('/^Contact:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/Contact:
> /ig');
> }
> Any tips?
> On Mon, Feb 7, 2011 at 11:19 AM, Tyler Merritt  wrote:
>>
>> I have used "if (status=="100")" and "if (is_method("TRYING"))" and placed
>> these in a variety of places, at the top of route[0], in route[1], in
>> onreply_route (where it actually works), but nowhere can I get the OUTGOING
>> 100 TRYING packet to change...
>> I can't figure this out - there isn't very good documentation surrounding
>> this particular packet, and if I were to guess, I would say that the method
>> is hard-coded somewhere in the routing module.
>> Any pointers on where this packet might be modified before sending it to
>> the upstream provider would be welcome.
>
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>

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Re: [OpenSIPS-Users] Help with is_present_hf

[Dave Singer]

Tyler,

Not sure but I believe custom headers are supposed to start with "X-"
and it may be that opensips is rejecting putting it in because it is
not a standard SIP header and it does not start with "X-".
Turning up the debug would probably show why.

Dave

On Mon, Feb 7, 2011 at 6:02 AM, Tyler Merritt  wrote:
> Using Opensips 1.6 ->
> Can anyone tell me what I'm doing wrong here?
> if (is_present_hf("P-Called-Party-Id")) {
>                          append_hf("GW: EPSILON\r\n");
>                          $avp(oriUri) =
> $(hdr(P-Called-Party-Id){s.substr,5,12});
>                          xlog("L_INFO", "Reply user now equals $avp(oriUri)
> from P-Called-Party-Id");
>   }
>
> Here is the packet that should hit this block:
> E.@...u.INVITE sip:NUMBER@HOST:5060 SIP/2.0
> Via: SIP/2.0/UDP REMOTEHOST:5060;branch=z9hG4bKbkr26a0010bgsbs5b640.1
> Max-Forwards: 66
> From: "NUMBER" ;tag=SDd4l3002-as19e30264
> To: 
> Contact: ;nat=yes
> CSeq: 103 INVITE
> P-Called-Party-Id: 
> Allow: INVITE, ACK, CANCEL, BYE
> As you can see - there is definitely a P-Called-Party-Id...
> The daemon.log actually shows the avp being set:
> /usr/sbin/opensips[13365]: Reply user now equals NUMBER from
> P-Called-Party-Id
> That's the only place in my config where I'm setting the $avp at all.
> And yet - I've added a bunch of header fields that help me watch where a
> packet routes through the code:
> GW: ALPHA
> GW: CHARLIE
> GW: NOVEMBER
> GW: SIERRA
> GW: TANGO
> GW: WILCO
> GW: XRAY
> GW: ZULU
>
> The header "GW: EPSILON" should be added if the packet were routed through
> that block of code - and it's not...
> I don't get it.  The functionality that I'm aiming for isn't working
> (resetting the outbound Contact to the inbound URI of the original request -
> but there's an Asterisk box in the way so it's the same as referencing two
> requests prior).
>
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Re: [OpenSIPS-Users] trying to rewriteuri to the value of a variable

[Dave Singer]

Isn't
   $var(my_uri) = "5...@example.com";
   $ru = "sip:" + $var(my_ruri);
the same as
   rewriteuri("sip:5...@example.com");
just that using $ru you can use it just like assigning to other vars
like you are doing.


On Sun, Feb 6, 2011 at 6:36 PM, Duane Larson  wrote:
> Avpops
>
> Sent from Droid
>
> On Feb 6, 2011 8:34 PM, "Nigel Daniels" 
> wrote:
>> the problem is that rewriteuri() does not support variables is there any
>> alternative to rewriteuri() for changing the uri
>>
>> On Sun, Feb 6, 2011 at 6:26 PM, Tyler Merritt  wrote:
>>
>>> I do this as well:
>>>
>>> $var(origFromNumber) = $(re{uri.user});
>>> $var(finalFromURI) = "sip:81" + $(var(origFromNumber){s.substr,1,0}) +
>>> "@"
>>> + $fd;
>>>
>>> This works fine for me. Takes what is essentially a DID and formats it to
>>> a SIP PEER string as per the carrier.
>>>
>>> You say that var(z) prints correctly in the logs, but according to the
>>> parser it's not a legit URI. It may just be that your upstream routes
>>> don't
>>> recognize this format.
>>>
>>> We use the usr_loc and db_aliases in order to match the URI up with a
>>> destination - have you ensured that there is a valid destination for this
>>> SIP URI?
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>> Nigel Daniels
>> Network & Systems Administrator
>> ConnectAndSell inc.
>> (650)-533-2542
>
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>
>

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Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu

[Dave Singer]

Tyler,

Just went through the OpenSIPS default script webminar =>
http://www.opensips.org/html/docs/video/webinar005/
And while the audio at the beginning is bad (and very end), it is only
just a little bit and it is because it was coming through a bad
connection to the seminar where the webinar was recorded.
If there truely is a problem with some of them try downloading them
instead of using the browser streaming. Also list which one(s) you
have trouble with.

Dave

On Thu, Feb 3, 2011 at 4:01 PM, Tyler Merritt  wrote:
>
> Dave,
> The audio on some of the webinars that I have watched has been almost 
> unintelligible :(  I like webinars - I present many of them in my work for 
> our customers, but I couldn't really hear well.
> I can't attend the live webinars as I'm in Tokyo - they happen at like 3 am.
> Anyway to clean up the audio?  Bogdan - can I send you a mic better mic :)
>
>   Tyler Merritt. Sales Engineer.
>   Contact: tmerr...@fonality.com | 310.861.4300 x 8850 |
>   fonality.com | SE Blog

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Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu

[Dave Singer]

The best place to start is http://www.opensips.org/
In the left column of the web page there is a section titled Resources
with links to many very helpful resources. Your using the mailing list
so you probably already have seen them to get here.
So. Where are you getting stuck? We need specifics in order to help out.

Also when you have a question you should start your own thread and not
use an existing thread unless it is completely relevant to what your
asking/stating.

FYI: The webinars are VERY important for getting an understanding of
how the whole thing works. With SIP the big picture is very important!
With out them you'll learn a lot of things the hard way like I did
before they were available.
Another good way to learn is to follow the mailing list discussions.

Welcome to the club,  ;-)
Dave

P.S. The software, documentation, mailing list, IRC, etc are all free
resources. The people helping you out are not getting paid to do it.
So an attitude of appreciation with patience will get you the best
millage. If you need more support there are those willing to do
contract support. See http://www.opensips.org/Resources/Business

On Wed, Feb 2, 2011 at 7:16 PM, Pradeep Patil  wrote:
> Anyone can help please in installing Opensip on Ubuntu.
>
>
>
>
>
> On Wed, Feb 2, 2011 at 1:15 AM, Duane Larson  wrote:
>>
>> The first thing you should do is
>> http://www.packtpub.com/article/installation-of-opensips-1.6
>>
>> You can watch the webinars here
>> http://www.opensips.org/Resources/Webinars
>>
>> You should join the mailing list
>> http://www.opensips.org/Resources/MailingLists
>>
>> To search old mailing list posts I use
>>
>> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>>
>> Sounds like what you need to do is to actually create a user/subscriber so
>> that opensips can register the x-lite client.  For that you need to use the
>> opensipsctl command or the osipsconsole.
>>
>>
>>
>> On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra 
>> wrote:
>>>
>>> Guys I a newbie to OpenSIPS
>>>
>>>  I have installed opensips and mysql on ubuntu following some
>>> instructions. I have also installed x-lite. Now how to register a user in
>>> opensips and to use it with the client ? I am stuck, please let me know
>>> Regards
>>> Ricky
>>>
>>> ___
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>>>
>>
>>
>>
>> --
>> --
>> *--*--*--*--*--*
>> Duane
>> *--*--*--*--*--*
>> --
>>
>> ___
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>>
>
>
>
> --
> thanking you,
> Pradeep Patil
> Cell No: 9676206432
>
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Re: [OpenSIPS-Users] How to test if a message is from myself

[Dave Singer]

Wow I missed that one. Thanks.
Does that work for PVs so I can test other IPs like one from another
header, say "X-src-ip:192.168.0.5". Last I tried I couldn't get it to
work. Not sure if that was 1.6.2 or 1.6.3. I'm using 1.6.4 now. :)

Thanks Again
Dave

On Wed, Feb 2, 2011 at 4:37 AM, Bogdan-Andrei Iancu  wrote:
> Hi Dave,
>
> do :  if (src_ip==myself) {}
>
> Regards,
> Bogdan
>
> Dave Singer wrote:
>>
>> Is there any way to check if the source IP/port is one that opensips
>> is listening on or one ? something like if ("sip:$si:$sp" == myself) {
>> ...bla; bla;}
>>
>> Thanks
>> Dave
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> OpenSIPS solutions and "know-how"
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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[OpenSIPS-Users] How to test if a message is from myself

[Dave Singer]

Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if ("sip:$si:$sp" == myself) {
...bla; bla;}

Thanks
Dave

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Re: [OpenSIPS-Users] Installing OpenSIPS on Red Hat

[Dave Singer]

Toyima,

I posted in the list last week on the thread "multiple
use_media_proxy() calls" my notes on getting media proxy installed on
centos 5.5.

To compile opensips just do the yum installs mentioned in that thread,
get the opensips source, unpack, compile and install according to docs
on install from source.
what I do after the yum installs and updates:

cd /usr/local/src/
wget "http://opensips.org/pub/opensips/latest/src/opensips-1.6.4-tls_src.tar.gz";
tar -xzf opensips-1.6.?-tls_src.tar.gz
cd opensips-1.6.?-tls
# note: the -j is the number of threads the compile can use. I
usually use one less than number of cores.
TLS=1 make -j3 prefix=/opt/opensips doc-prefix=/usr/local
man-prefix=/usr/local include_modules="db_mysql db_postgres" install

A few months ago I also posted an init.d script for centos that does
double checking for problems (runs opensips with -c and runs it full
on localhost:2021 or some port) because -c only does syntax checking
and I got tired of restarting and it would fail when starting back up.
It also is set to use monit if it is installed and configured.

Dave

On Mon, Jan 31, 2011 at 11:23 AM, Brett Nemeroff  wrote:
> On Mon, Jan 31, 2011 at 10:28 AM, Toyima Dias  wrote:
>>
>> Hello,
>>
>> I've seen many information on how to install OpenSIPS in DEBIAN, but in my
>> case i need to install it on Red Hat, i've found some information but not
>> very well supported, i would like to have a very clean and stable
>> installation for a very large and stable production server, is there any
>> recommendations please?
>>
>> Regards
>
> Toyima,
> If you are trying to install mediaproxy, then yes, Debian is the way to go.
> With it's dependencies you'll be fighting to make it work for a while.
> However, if you are just trying to get Opensips to run on Redhat.. Can't you
> just compile from source? It's really not so advanced. Maybe I'm missing
> something?
> -Brett
>
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Re: [OpenSIPS-Users] OpenSIPS handling B2B features

[Dave Singer]

Toyima,

Asterisk is a pain for making conference work because of its
dependence on the dahdi kernel driver for timing.
I like the FreeSwitch implementation. FreeSwitch has a very active irc
at irc.freenode.net.

Dave

On Thu, Jan 27, 2011 at 12:21 AM, Toyima Dias  wrote:
>
> Anca,
>
> What conference system would you recommend for me? Asterisk? SEMS? may be 
> some licensed software could be ok for me...i would like to heard some 
> recommendations
>
> Regards
>
> 2011/1/26 Anca Vamanu 
>>
>> Hi Toyima,
>>
>> On 01/25/2011 06:38 PM, Toyima Dias wrote:
>>>
>>> Thanks Anca,
>>> To make transfers possibles, i would use REFER messages, and for this i 
>>> would need to use B2BUA module of opensips, rigth?
>>
>> VoIP phones can handle REFER messages, the problem is with the gateways, and 
>> this is why sometimes it is required to handle the REFER at the server and 
>> instead of forwarding it, replace it with connecting the party in a call to 
>> the new party from the server.
>>>
>>> What about conferences? For conferences is the same procedure (REFER 
>>> messages), but always handling a media b2bua, rigth?
>>
>> For conferences it is not enough to handle the signaling, you also need a 
>> conference bridge to distribute the media to more parties.
>>
>> Regards,
>>
>> --
>> Anca Vamanu
>> www.voice-system.ro
>>
>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] multiple use_media_proxy() calls

[Dave Singer]

I have 2.4.4 running on centos 5.5. Here are my notes for getting it working
using mostly rpms.
Hope this helps you out and others too. If anyone wanted to put this in an
official doc for installing mediaproxy, that would be cool or point me to
where I could do that. :)

install stuff for opensips (plus some essential tools like vim and screen)
install that some is also used for mediaproxy. This is from a centos
minimal/barebones install point:

rpm -Uhv "http://yum.pgsqlrpms.org/reporpms/8.4/pgdg-centos-8.4-2.noarch.rpm
"
# had to use wget because rpm couldn't pull the url for some reason,
probably a redirection.
wget "
http://packages.sw.be/rpmforge-release/rpmforge-release-0.5.2-2.el5.rf.x86_64.rpm
"
rpm -Uhv rpmforge-release-*.rpm
rpm -Uhv "
http://download.fedora.redhat.com/pub/epel/5/x86_64/epel-release-5-4.noarch.rpm
"
yum install -y yum-fastestmirror yum-priorities vim-enhanced man rsyslog
wget screen
yum install -y vixie-cron monit rsyslog logrotate rsync tcpdump wireshark
ngrep patch postgresql-server mysql-devel postgresql-devel bison lynx flex
openssl-devel libxslt subversion make gcc wget vim-enhanced screen
which libxml2-devel
pcre-devel db4-devel net-snmp-devel expat-devel libmemcache-devel mysql-server
mysql
sed -r -i '/^\[/apriority = 20' /etc/yum.repos.d/epel*
sed -r -i '/^\[/apriority = 30' /etc/yum.repos.d/rpmforge.repo
sed -r -i '/^\[/apriority = 10' /etc/yum.repos.d/CentOS-*
sed -r -i '/^\[/apriority = 5' /etc/yum.repos.d/pgdg-84-centos.repo

Mediproxy install:
put this together primarily using the links below
http://voiprookie.blogspot.com/2009/04/blog-post.html
http://www.smartvox.co.uk/serfaq_install_mediaproxy2.htm
CentOS 5.5  install prerequisites:
NOTES:
  DO NOT install twisted or gnutls stuff with yum!! remove if rpm is
installed installed. ( rpm -qa | grep -E "twisted|gnutls" )
  Some newer versions of source packages below may not work because of
conflicts with CentOS versions of stuff like the kernel.

yum install libnetfilter_conntrack libnetfilter_conntrack-devel
python-dns python-ctypes pyOpenSSL python-application python-cjson
python-devel python-sqlobject python-zope-interface

wget "ftp://ftp.gnupg.ca/gnutls/gnutls-2.4.2.tar.bz2";
tar -xjf gnutls-2.4.2.tar.bz2
cd gnutls-2.4.2
##./configure --with-included-libtasn1 --libdir=/usr/local/lib64
./configure --with-included-libtasn1 --libdir=/usr/lib64
make
make install
cd /usr/local/lib64/
##for i in libgnutls.so.26 libgnutls.so.26.4.6 libgnutls-extra.so.26
libgnutls-extra.so.26.4.6 libgnutls-openssl.so.26
libgnutls-openssl.so.26.4.6 ; do cp -v $i /usr/lib64/ ; done
cd /usr/local/src/
wget "
http://pypi.python.org/packages/source/p/python-gnutls/python-gnutls-1.1.9.tar.gz#md5=bdd81a9dd3c2f87393cd6c014bf6885b
"
tar -xzf python-gnutls-1.1.9.tar.gz
cd python-gnutls-1.1.9
./setup.py build
./setup.py install
cd ..
wget "http://tmrc.mit.edu/mirror/twisted/Twisted/8.1/Twisted-8.1.0.tar.bz2";
tar -xjf Twisted-8.1.0.tar.bz2
cd Twisted-8.1.0
./setup.py build
./setup.py install

setup mediaproxy
wget "http://download.ag-projects.com/MediaProxy/mediaproxy-2.4.3.tar.gz";
tar -xzf mediaproxy-2.4.3.tar.gz
mv mediaproxy-2.4.3 /opt/mediaproxy
cd /opt/mediaproxy
./build_inplace
./media-relay --no-fork
lsmod
/etc/init.d/iptables status
/etc/init.d/iptables start
/etc/init.d/iptables status
modprobe ip_tables
lsmod
./media-relay --no-fork
./media-dispatcher --no-fork

sysctl -w net.ipv4.ip_forward=1
sed -i -r
's/^.*\
relay_ip = 
port_range = 2:6
log_level = ERROR
; log_level options ( DEBUG, INFO, WARNING, ERROR and CRITICAL)
[Dispatcher]
socket_path = /var/run/mediaproxy-dispatcher.sock
management_use_tls = no
listen = :5500
log_level = ERROR
; log_level options (DEBUG, INFO, WARNING, ERROR and CRITICAL)
[OpenSIPS]
socket_path = /tmp/opensips_fifo
max_connections = 10
Opensips.cfg Stuffopensips modules to load in opensips.cfg:
loadmodule "dialog.so" # requred for mediaproxy_engage and kill of (dead
calls)
loadmodule "mediaproxy.so" # if using mediaproxy_engage it is the only
mediaproxy function used in the script and only for initial INVITE.
loadmodule "mi_datagram.so" # required for mediaproxy to tell opensips to
end a call that rtp has timed out for (dead calls).
loadmodule "nathelper.so" # needed to fix sdp origin header to hide topoloty
that media doesn't fix but should. -eg- fix_nated_sdp("8","208.76.155.250");
# with ip of rtp relay, can be var. use in invite and replies with sdp -
detect sdp with if ( $cl > 0 )

params:
modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
modparam("mi_datagram", "children_count", 4)
modparam("dialog", "dlg_flag", 13)
# db_mode: 0 - NO_DB, 1 - REALTIME, 2 - DELAYED, 3 - ON SHUTDOWN
modparam("dialog", "db_mode", 3)
modparam("dialog", "db_url","postgres://opensips:passwd@localhost/opensips")
modparam("mediaproxy", "mediaproxy_socket",
"/var/run/me

Re: [OpenSIPS-Users] CDR Accounting

[Dave Singer]

Thanks for that clarification!

So just to be sure I'm clear on this. $Ts rounds down ( truncates ) the
current second. So using $avp(s:start_time) = $Ts.$Tsm; would give something
like "12343253.543233" and always be accurate?
Further (standard ACC [without dialog]) you could just put $Tsm in the
db_extras as an extra column and then use the tool you are using to created
cdrs from the acc table to take the column with the $Tsm value into account
when calculating duration, ringtime, etc.
To get these numbers with the new CDR feature, you would have to store a
dialog variable like $dlg_val("start") = "$Ts.$Tsm" at start
and $dlg_val("answered") = "$Ts.$Tsm" at answer and at the end of the call
add them db_extras along with $Ts.$Tsm for the end of the call. You would
then want to go off of those numbers rather than what the acc module
automatically put in the db.
The disadvantage would be having duplicate columns for start/answer/end
times but you would still have the benefit of not having to have an external
process building cdrs from acc table matching up BYEs to INVITES, etc.
Calculating the durations in opensips would be tricky since it does not work
with non integers. But you could do that easily with a sql trigger on the
record insert.

Does this sound right?

Thanks
Dave

On Thu, Jan 20, 2011 at 4:54 AM, Anca Vamanu  wrote:

> On 01/19/2011 09:31 PM, a.zagors...@oyster-telecom.ru wrote:
>
>> It was my conversation, and I told about
>> 1) $Tsm is MICROseconds (6 digits)
>> 2) It is microseconds of a current second.
>>
>>  Yes, that's true. I fixed it in documentation also.
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
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Re: [OpenSIPS-Users] Handling messages to more than one final destination

[Dave Singer]

Anca,

When using t_replicate, is opensips then expecting a response from the
server it is replicated to? I'm guessing not and also therefore will not
send retries when/if no response is received. Then on the recording server
you can just use tcpdump to save the messages.
Further you would need to t_replicate every message, catching it before any
t_relay (actually replacing t_relay) including loose routing at the
beginning of the script and also in on_reply routes.
Are these assumptions correct?

A note on many destinations, I've been using it for replicating
registrations to voicemail servers and had to use append_branch to my other
destinations and t_replicate to the last one. I'm not sure how it then deals
with where it expects a response from if not all destinations (depending on
answer to above question).

Thanks
Dave

On Mon, Jan 24, 2011 at 9:26 AM, Anca Vamanu  wrote:

>  On 01/24/2011 07:19 PM, Toyima Dias wrote:
>
> Thanks Anca,
>
> I can send the call as many destinations i want, right? as many
> t_replicate, many destinations? but now i have a doubt, is it right what i
> want to do? i mean: sending the INVITE to both the final destination and the
> recorder server to record conversations between peers? should i treat the
> recording server as a media server right? why not handling this recording
> server like a voicemail server with rewritehostport?
>
>
> For the voicemail server - it is the last destination of the call, this is
> why you can use rewritehost port. But for the use case that you described,
> recording the SIP messages on another server, t_replicate is what you need.
>
> Regards,
> Anca
>
>
>  Thanks in advance for your help
>
>
>
>
>
> 2011/1/24 Anca Vamanu 
>
>> Hi Toyima,
>>
>> You can use t_replicate function:
>> http://www.opensips.org/html/docs/modules/devel/tm.html#treplicate.
>>
>> Regards,
>>
>> --
>> Anca Vamanuwww.voice-system.ro
>>
>>
>
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Re: [OpenSIPS-Users] CDR Accounting

[Dave Singer]

Be careful when using $Tsm. I read a thread the other day that was just
talking about it and that it returns milliseconds since midnight, not epoch.
So if that is true you will need to somehow handle calls that cross midnight
for duration at least and start/answer/end of call if including the
precision in those timestamps.
I'd be interested to see what you find.
What I've seen with sub second accounting is for rounding purposes where for
example billing in 6 second increments and it is always rounded up. So a
call 6.001 seconds would be charged for 12 seconds.

Dave

On Wed, Jan 19, 2011 at 6:35 AM, Brett Nemeroff  wrote:

> On Wed, Jan 19, 2011 at 8:32 AM, Andrew Philp 
> wrote:
>
>> We are looking to use this for billing directly and for most of our
>> customers are billed on a hundredth of a second.
>>
>> Does this make sense?
>>
>>
> Sure it does, but I've never heard of sub second billing. The ACC module is
> probably not going to help you for this need presently. I believe $Tsm
> returns milliseconds. You may be able to manually account for the calls with
> that.
> -Brett
>
>
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Re: [OpenSIPS-Users] Some strange things with starting opensips

[Dave Singer]

Adjust STARTOPTIONS to include "-p /var/run/opensips.pid" ( replace with the
path to where you want the pid file )

There are other options you may want to adjust for starting opensips. Note
that some options are overridden if set in the config file like listen
address. Here is the list of options you can see when you run opensips with
the help option -h
/sbin/opensips -h
version: opensips 1.6.4-tls (x86_64/linux)
Usage: opensips -l address [-l address ...] [options]
Options:
-f file  Configuration file (default
/opt/opensips/etc/opensips/opensips.cfg)
-c   Check configuration file for errors
-C   Similar to '-c' but in addition checks the flags of
exported
  functions from included route blocks
-l address   Listen on the specified address/interface (multiple -l
  mean listening on more addresses).  The address format is
  [proto:]addr[:port], where proto=udp|tcp and
  addr= host|ip_address|interface_name. E.g: -l locahost,
  -l udp:127.0.0.1:5080, -l eth0:5062 The default behavior
  is to listen on all the interfaces.
-n processes Number of child processes to fork per interface
  (default: 8)
-r   Use dns to check if is necessary to add a "received="
  field to a via
-R   Same as `-r` but use reverse dns;
  (to use both use `-rR`)
-v   Turn on "via:" host checking when forwarding replies
-d   Debugging mode (multiple -d increase the level)
-D   Do not fork into daemon mode
-E   Log to stderr
-T   Disable tcp
-N processes Number of tcp child processes (default: equal to `-n`)
-W methodpoll method
-V   Version number
-h   This help message
-b nrMaximum receive buffer size which will not be exceeded by
  auto-probing procedure even if  OS allows
-m nrSize of shared memory allocated in Megabytes
-w dir   Change the working directory to "dir" (default "/")
-t dir   Chroot to "dir"
-u uid   Change uid
-g gid   Change gid
-P file  Create a pid file
-G file  Create a pgid file

Dave

On Fri, Jan 14, 2011 at 7:20 AM, Chris Liu  wrote:

> Hi Anca,
>
> Thank you very much , i add this line to
> /etc/opensips/opensipsctlrc
>
> STARTOPTIONS=" -f /etc/opensips/opensips.cfg"
>
> opensipsctl start use this configure file now. Thanks
>
> Do you know why I start with /etc/init.d/opensips start  , there is no
> opensips.pid in /var/run ?
>
> Regards,
>
> Chris
>
>
> --
> *From:* Anca Vamanu 
> *To:* users@lists.opensips.org
> *Sent:* Fri, January 14, 2011 9:37:59 AM
> *Subject:* Re: [OpenSIPS-Users] Some strange things with starting opensips
>
> Hi Chris,
>
> You need to set the path to the configuration file.
> You should edit file opensipsctlrc ( localed in etc/opensips/ ) and set
> STARTOPTIONS=" -f /etc/opensips/opensips.cfg" ( replace with the path to
> your configuration file).
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
>
> On 01/14/2011 06:26 AM, Chris Liu wrote:
>
>  Hi everyone,
>
> We can use opensipsctl start and service opensips start to startup
> opensips, can anybody show me the different of these two ways?
>
> On my CentOS system, when I use service opensips start tostartup the
> system, no opensips.pid in /var/run and I can find /tmp/opensips_fifo but it
> will disapear in about 10 mins. If I use opensipsctl start to start the
> system, /tmp/opensips_fifo and opensips.pid are fine.
>
> But interest thing is seems service opensips start doesn't use the
> configure file. I changed the name of configure file and try service
> opensips start, It works with no error! Anybody know which configure file
> be use by it?
> [root@OpenSIP1/etc/opensips]# mv opensips.cfg opensips.cfg_
> [root@OpenSIP1/etc/opensips]# opensipsctl stop
> INFO: Stopping OpenSIPS :
> INFO: stopped
> [root@OpenSIP1/etc/opensips]# opensipsctl start
> INFO: Starting OpenSIPS :
> INFO: started (pid: 15088)
>
>
>
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[OpenSIPS-Users] solution to opensips control pannel MI connection using fifo on CentOS 5 - selinux audit2allow

[Dave Singer]

Banged my head for a while with why I couldn't get fifo working for the
Control Panel
I was getting "sorry -- cannot open write fifo".

Hope this can help other and maybe even make it into the docs.
Found two problems.

1. Apache process couldn't use /tmp/opensips_fifo because the permissions
were prw-rw 1 root root
I fixed that temporarily by chmod o+rw /tmp/opensips
Unfortunately restarting opensips voids that fix unless you do chmod after
each start after the fifo is created
So I changed the user opensips runs as using the -u apache -g apache
options.
This solved my problem and made my system more safe from hackers through sip
messages.

2. SELinux (hate the good thing) was also blocking apache using it.
I did some googling and found some mention of disabling selinux. Tried that
and things worked.
So I tried that and whala! IT WORKED!!!
Now I was bummed cause I don't want to completely disable SELinux, pain
though it is.
So I finally broke down and looked into the docs on for SELinux and found
audit2allow. YEA
All you have to do to get SELinux to permantly allow something that should
happen is pipe the audit lines that show the problem from
/var/log/audit/audit.log (or /var/log/messages)
Since an app my stop trying things SELinux may block after an initial
rejection you want to set SELinux to permissive with
  setenforce Permissive
Perform the actions like us the MI from opensips-cp stopped by selinux
before then you can use the logs to allow those specific actions.
I used less to find the specific lines then copied them into a file with
just those lines.

Here is an example with a file that only has the audit logs I want to allow:
cat selinux_blocked_my_app | audit2allow -M opensipscp
semodule -i opensipscp.pp

That is it!!
I wanted to transport it to other servers and make it easy to do on an
install so I dug a little more and came up with this I can just cut and
paste:

cat > opensipscp.te <___
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Re: [OpenSIPS-Users] [NEW] Media timeout detection and call termination

[Dave Singer]

It seems VAD would also be a potential problem with NAT and state
full packet inspection on firewalls. It would see a better practice for the
UA to send keep alive RTP packets. That would solve both dead call detection
and potential NAT issues. That would be nice RFC change.
In my situation I have never seen VAD in use. I would like to have the
option to set what determines a call is dead, 1 leg no RTP, both legs no
RTP, or don't kill a call no matter if there is no rtp.
Looking forward to the solution you are talking about.

Dave
On Fri, Dec 17, 2010 at 9:48 AM, Razvan Crainea
wrote:

> Hi Saul,
>
> Unfortunately this solution doesn't deal with the UA using VAD in case
> there are no RTP packages sent when voice is suppressed. Anyway, we are
> working on a solution that will solve this problem.
>
> Regards,
>
>
>
> On 12/17/2010 05:02 PM, Saúl Ibarra Corretgé wrote:
>
>>
>>  As OpenSIPS is a proxy and at signaling level there is no continues
>>> traffic to help in detecting the ghost calls, such detection must be
>>> done at media level.
>>>
>>>
>> Hi,
>>
>> If this detection is done based only in the lack of RTP, then you could
>> say that a UA that uses VAD (it doesn't transmit RTP when there is silence)
>> timed out, but what really happened is that it was silent for a long time.
>> How is the timeout condition detected?
>>
>> Did this new addition deal with this situation?
>>
>>
>> Thanks and regards,
>>
>>
> --
> Razvan Crainea
> www.voice-system.ro
>
>
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Re: [OpenSIPS-Users] Freeswitch vs Asterisk

[Dave Singer]

We have both asterisk and Freeswitch in production. The primary place where
we have * installed is as a pbx for our business customers (where we started
doing business and didn't know any better). We are still using * for them
for two reasons: migration time and voicemail app I feel is still better in
a couple points. They are low volume usage so crashes are very rare.
We also have some boxes where we connect to telecom PRI circuits where the
API for FS doesn't support some params we need to set. So we are stuck there
for now. There systems handle moderate volume, 30 - 90 simultaneous calls.
This call volume has proved to be deadly to asterisk and we have to restart
asterisk daily or suffer a crash in the middle of peek times.
We use FreeSwitch as the workhorse with a custom routing module combined
with Opensips as a class 4 switch (whole sale trunking service). With high
powered servers (latest dual xeon quad core, 16GB ram, and 10Gbit ethernet)
it can handle thousands of simultaneous calls. They run for months without
problem (would be longer but for reboots for upgrades, etc., not FS
crashes).
We also have a class 5 system that handles residential users which uses FS
and opensips for failover. Again no FS crashes.
FS is also our conference server for all our services.

We started out using * building the business PBXs. Later found FS as we were
developing the residential system and converted to using it.
Coming from * to FS has some difficulties because of the different ways of
doing things like the flow of the dialplan where all conditions are
evaluated at the time of entry to the dialplan, not as each line is executed
(executing another extension solved this problem for me).
I do think FS has a little higher learning curve, I have found it better in
almost every area, especially stability and flexibility.

Well, those are my 2 cents. :-D
Dave

On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins  wrote:

> Comments inline. (Full disclosure: I am on the FreeSWITCH team, so if I
> come off as biased then you know why. ;)
>
> On Tue, Dec 7, 2010 at 8:29 AM, paul.gor...@gmail.com <
> paul.gor...@gmail.com> wrote:
>
>> We use freeswitch in prod alone, no opensips yet. I would say fs is
>> definetly more scalable than *.
>> Stability wise seems like fs is on par with *.
>>
> YMMV, but a large percentage of FreeSWITCH users have abandoned Asterisk
> specifically because of stability issues, like random and inexplicable
> crashes.
>
>
>> * has substantially better interface for control over socket connection -
>> it's easier to implement and it's more consistent.
>>
> This statement is patently false. The FreeSWITCH event socket interface is
> incredibly powerful and is absolutely more consistent than the AMI. Those
> wondering about inconsistencies in the AMI should listen to a seasoned AMI
> developer talk about the challenges:
> http://www.viddler.com/explore/cluecon/videos/29/
>
>
>> Configuration wise, I think * is easier, xml- based approach in fs is
>> cumbersome and has no real advantage over *.
>>
> This one really is like Coke vs. Pepsi. Some people hate XML, some people
> hate INI-style config files. Personally, I've done both and now that I'm
> accustomed to FreeSWITCH's XML files I find them much easier to read than
> Asterisk's config files. There is one "real advantage" to using XML for
> configs and that is that machines and humans can both produce XML, so it's
> relatively simple to let a machine generate XML-based configs on the fly.
> (FreeSWITCH uses "mod_xml_curl" as the basis for dynamic configuration -
> it's very cool and I recommend that you check it out.)
>
>
>> We have endless problems with fs nat handling, lots of no audio issues
>> with end users behind a nat. That's why we want to try opensips solution for
>> that.
>>
> Almost all NAT problems stem from phones which don't handle NAT properly or
> NAT devices that scramble ports and IP addresses when packets pass through.
> FreeSWITCH has several NAT-busting tools to assist the system admin. Some
> tools are for when FS is behind NAT, others are for when the phones are
> behind NAT. Bottom line is this: if the NAT device and the phones are not
> horribly broken then FS works great with NAT and in many cases "just works."
> However, when you start mixing crazy scenarios with broken phones then bad
> things will happen. Example: Polycom phones are wonderful except that they
> don't support rport - FS has a mechanism to assist with this but if you turn
> it on to "fix" the Polycom phones then it will break all other phone types.
> (There is a limit to the amount of pandering that the FS devs will do in
> order to interop with broken devices. In many cases they simply say "NO" to
> doing stupid things in order to work with broken devices. If you must work
> with such a device then perhaps FreeSWITCH isn't for you.)
>
> All that being said, the FreeSWITCH developers have a simple mantra that
> they follow to the letter: Use what works for 

Re: [OpenSIPS-Users] mediaproxy 2.4.2

[Dave Singer]

I replaced the 5060 with 5065, recompiled (make clean; make) and renamed the
binary to pcapsipdump5065 so I would know which one I was using. Worked
great.

Dave

On Fri, Dec 3, 2010 at 8:28 AM, Saúl Ibarra Corretgé
wrote:

> On 12/03/2010 04:50 PM, Iñaki Baz Castillo wrote:
>
>> 2010/12/3 Saúl Ibarra Corretgé:
>>
>>> Thanks for sharing :-) You may also find pcapsipdump
>>> (http://sourceforge.net/projects/pcapsipdump/) useful.
>>>
>>
>> I invite you to use pcapsipdump when you rproxy listens in a port
>> different than 5060:
>>
>> /usr/src/pcapsipdump-0.1.4# grep -r 5060 *
>> pcapsipdump.cpp:}else if
>> (htons(header_udp->source)==5060||
>> pcapsipdump.cpp:htons(header_udp->dest)==5060){
>>
>>
>>  Wow, what a failure! Thanks for pointing it out :-)
>
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Handle end of a transaction/dialog

[Dave Singer]

I'm not shure exactly when the automatick removal from groups is, you would
have to expiriment.
There are only a few end of call situations to cover.
   Bye after a call was accepted (200 OK to INVITE). That comes in main
routing block but would usually be forwarded in the loose_route section. Add
an if (is_method("BYE")) and put logic there.
   Cancel. That also comes in main routing block and is usually handled just
below the has_totag() stuff. Find is_method("CANCEL"). You would add it to
the if (t_check_trans())
   Failures. Handled in the failure route block you activated for the call
with t_on_failure("?")
In each section, just use the unset_dlg_profile function for the group it
would be in.
Be careful with the BYE as it can come from either side of the call and
depending on what your grouping formula is based on you might need to make a
special case depending which end the BYE came from. I haven't looked at BYE
packets much. You may want to use
store_dlg_value(name,val)<http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294530>
to
sore the group to pull it out at BYE with
fetch_dlg_value(name,pvar)<http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294595>
.

On Fri, Nov 19, 2010 at 3:16 PM,  wrote:

> So, you explained how to store and grouping counters. Thanks.
> But how to catch the moment when the opensips transaction is going down? I
> need to react immediately.
>
> I can catch a BYE request when a established sip session is ending. What
> about other cases?
>
>
>
> On Fri, 19 Nov 2010 09:27:38 -0800
>  Dave Singer  wrote:
>
>> You can use a combination of the following dialog profiles (grouping
>> calls)
>> functions for this:
>> get_profile_size(profile,[value],size)<
>> http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294302>
>>
>> Check
>> if a prifile is at limit.
>> set_dlg_profile(profile,[value])<
>> http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294092>
>>
>> If
>> not at limit, increment the count for the profile (NOTE the value option
>> is
>> not actually supported yet)
>> unset_dlg_profile(profile,[value])<
>> http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294162>
>>
>> If
>> need to remove from a profile like carrier failover
>> is_in_profile(profile,[value])<
>> http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294231>
>>
>> Not
>> sure this would be needed but...
>>
>> When a call ends it will automatically be removed from any profiles it was
>> part of.
>>
>> On Fri, Nov 19, 2010 at 1:29 AM, Anton Zagorskiy <
>> a.zagors...@oyster-telecom.ru> wrote:
>>
>>  Nobody uses limitation on amount of established sessions?  I don't
>>> believe!
>>> =)
>>>
>>>
>>>
>>>
>>>
>>>
>>> WBR, Anton Zagorskiy
>>> VoIP Developer, Oyster Telecom
>>> Phone.: +7 812 601-0666
>>> Fax: +7 812 601-0593
>>> a.zagors...@oyster-telecom.ru
>>> www.oyster-telecom.ru
>>>
>>>
>>>
>>> > -Original Message-
>>> > From: users-boun...@lists.opensips.org [mailto:users-
>>> > boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
>>> > Sent: Thursday, November 18, 2010 4:58 PM
>>> > To: 'OpenSIPS users mailling list'
>>> > Subject: [OpenSIPS-Users] Handle end of a transaction/dialog
>>> >
>>> > Hello.
>>> >
>>> > I need to control how much SIP dialogs (established or pending) has
>>> > each
>>> > domain. If that value is equal to certain constant then I should drop
>>> > any
>>> > new INVITE from that domain.
>>> > There is no problem with catching initial requests, but with
>>> > terminating.
>>> > How can I control that an openSIPS transaction is in the terminating
>>> > stage?
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > WBR, Anton Zagorskiy
>>> > VoIP Developer, Oyster Telecom
>>> > Phone.: +7 812 601-0666
>>> > Fax: +7 812 601-0593
>>> > a.zagors...@oyster-telecom.ru
>>> > www.oyster-telecom.ru
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > ___
>>> > Users mailing list
>>> > Users@lists.opensips.org
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>
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Re: [OpenSIPS-Users] Handle end of a transaction/dialog

[Dave Singer]

You can use a combination of the following dialog profiles (grouping calls)
functions for this:
get_profile_size(profile,[value],size)
Check
if a prifile is at limit.
set_dlg_profile(profile,[value])
If
not at limit, increment the count for the profile (NOTE the value option is
not actually supported yet)
unset_dlg_profile(profile,[value])
If
need to remove from a profile like carrier failover
is_in_profile(profile,[value])
Not
sure this would be needed but...

When a call ends it will automatically be removed from any profiles it was
part of.

On Fri, Nov 19, 2010 at 1:29 AM, Anton Zagorskiy <
a.zagors...@oyster-telecom.ru> wrote:

> Nobody uses limitation on amount of established sessions?  I don't believe!
> =)
>
>
>
>
>
>
> WBR, Anton Zagorskiy
> VoIP Developer, Oyster Telecom
> Phone.: +7 812 601-0666
> Fax: +7 812 601-0593
> a.zagors...@oyster-telecom.ru
> www.oyster-telecom.ru
>
>
>
> > -Original Message-
> > From: users-boun...@lists.opensips.org [mailto:users-
> > boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
> > Sent: Thursday, November 18, 2010 4:58 PM
> > To: 'OpenSIPS users mailling list'
> > Subject: [OpenSIPS-Users] Handle end of a transaction/dialog
> >
> > Hello.
> >
> > I need to control how much SIP dialogs (established or pending) has
> > each
> > domain. If that value is equal to certain constant then I should drop
> > any
> > new INVITE from that domain.
> > There is no problem with catching initial requests, but with
> > terminating.
> > How can I control that an openSIPS transaction is in the terminating
> > stage?
> >
> >
> >
> >
> >
> >
> > WBR, Anton Zagorskiy
> > VoIP Developer, Oyster Telecom
> > Phone.: +7 812 601-0666
> > Fax: +7 812 601-0593
> > a.zagors...@oyster-telecom.ru
> > www.oyster-telecom.ru
> >
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] New to OpenSIPS

[Dave Singer]

Learning opensips is not like learning a programming language like C, ruby,
puthon, etc. It is very important to get an understanding of how sip flows
and opensips works (get the big picture) then get into the details of how to
manipulate things.
Checkout the past webinars on the Webinars
page
First two I'd look at are:
SIP introduction 
OpenSIPS default script 
You may want to get and start playing with the live
DVDavailable
that you can use in a virtual server or a spare computer without
installing.
Further, look at the
tutorialsfor specific
goals.
OpenSIPS Admin 
Course

The "Cookbooks " page is
basically the links to reference manual type documentation of functions,
features, variables etc. But it helps to have an understanding of how
opensips works from the suggested webinars.
Opensips IRC on freenode is quite active as well as the mailing list. Be
sure to ask questions there. But remember that like other opensource
software you are not paying for support so don't have an attitude of
expecting fast answers.

Also consider http://www.opensips.org/Training/EBootcamp.

I sure wish these resources were available when I first started with openser
(opensips) a few years ago. I have recently gone through some of these and
still found them helpful.

For performance check the Performance
tests page
and search the Mailing lists
 and Forums  for results users
have found.

Good luck
Dave (cando)

On Wed, Nov 17, 2010 at 11:54 AM, Austin Smith wrote:

> Greetings!
>
>
>
> I am new to opensips and am looking for a good place to start learning the
> application.
>
>
>
> I am curious if there is any recommended hardware or solid benchmarking
> statistics available somewhere.  Next I will need to learn on the
> configuration aspects of the software.  I am not looking for “cookbook”
> material, but rather language references and structure guidelines.
>
>
>
> Can anyone advise where these items may be available?
>
>
>
> Many thanks-
>
>
>
> Austin Smith,* **A+, NET+, SMBE, MCSA*
>
> *Director of Information Techology*
>
> *Digital Compass*
>
>
>
> (404) 410-2708 direct
>
> (404) 410-2701 fax
>
> 6 Concourse Parkway, #1500
>
> Atlanta, GA 30328
>
>
>
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>
>
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Re: [OpenSIPS-Users] Timer Based Failover Question

[Dave Singer]

I ran into the same problem with one of our carriers. The way I did it, with
advice from bogdan, was to set the fr_inv_timer_avp to 6 sec (so long
because of some calls to cell phone systems have long delays) when sending
to the particular carrier then in a reply route special to that carrier,
reset it to the normal 300 if the response was a 18x. Here are
the pertinent parts:

modparam("tm", "fr_inv_timer", 300) # Timer on Final response:
Minimum is 2 sec, Default is 120 sec.
modparam("tm", "restart_fr_on_each_reply", 1)   # Reset fr_int_timer on each
reply. Needed if you want to adjust the fr_inv_timer_avp with avp depending
on reply.
modparam("tm", "fr_inv_timer_avp", "$avp(i:2)") # Used if overide of
fr_inv_timer param is needed.
modparam("tm", "onreply_avp_mode", 1)   # set to 1 if you want to
access and or save avps from or for other parts of the transaction, like
changing the fr_inv_timer_avp.

route[carrier_c] {
if (is_method("INVITE")) {
t_on_failure("2");
t_on_reply("2");
$avp(i:2) = 6;
}
}
onreply_route[2] {
/* once we get ring progress let it ring for upto 300 sec */
fix_nated_contact();
if ( $rs =~ "18." ) {
$avp(i:2) = 300;
#xlog("got ringing, reset final timer to $avp(i:2) sec.\n");
}
}

On Tue, Nov 16, 2010 at 2:49 AM, Denis Putyato  wrote:

> And what about
>
> http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id250384
>
>
>
>
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Bruce Borrett
> *Sent:* Tuesday, November 16, 2010 1:40 PM
> *To:* Users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] Timer Based Failover Question
>
>
>
> Hi All
>
> I am having a problem where a SIP provider are sometimes sending us a 100,
> but then nothing afterwards. I would like to fail these calls over using a
> timer, but fr_timer wont work since we are receiving a 100, and fr_inv_timer
> requires a very lengthy duration which also will not work as I would like
> for the call to failover within 5 seconds maximum.
>
> Does anyone have any other suggestion for me please?
>
> Regards,
> Bruce Borrett
>
>
>
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Re: [OpenSIPS-Users] Mediaproxy broken

[Dave Singer]

I attached a graph of traffic on the interface that media-relay uses from
the 18th through 20th. Before that we didn't realize it wasn't working and
after we stopped it. FYI the interface has a 10Gb internet connection, so
graph numbers are not pushing internet limits.
The mediaproxy log compressed from 272MB to 11MB and is at this
link<http://viper.wideideas.net/mediaproxy.log.gz>
.

Thanks for any help.
Dave

On Wed, Oct 20, 2010 at 11:43 PM, Saúl Ibarra Corretgé  wrote:

> Hi,
>
> On 10/20/2010 11:14 PM, Dave Singer wrote:
> > I have one server where mediaproxy is having serious problems.
> > In the logs I have a lot of:
> > media-dispatcher[6067]: error: failed to end dialog: 500 Operation
> > failed
>
> This error is returned by OpenSIPS when MediaProxy calls the dlg_end_dlg
> over the MI interface. Do you see anything in the OpenSIPS log?
>
> > and a lot of of the following but much less than the above:
> > media-dispatcher[6067]: error: Got `remove' command from OpenSIPS
> > for unknown session with call-id `83335711-572b-122e-57ab-0024e869f3a5'
> >
>
> Are you using the engage_media_proxy function alone? Are you using
> end_media_session by any chance?
>
> > After some preiod of time, from a few minutes to a day or
> > so, media-relay appears to start holding open ports that have long gone
> > dead.
> > After a while it runs out of ports for new calls and opensips sends the
> > call on without using mediaproxy.
> > Traffic graphs support this conclusion. They also show that it was
> > working fine for more than a month then stopped working.
> >
>
> Can you please provide some more information/logs on this?
>
> > I'm using
> >opensips 1.6.2-notls (x86_64/linux)
> >mediaproxy is enacted with engage_media_proxy() with dialog write to
> > db on shutdown.
> >mediaproxy 2.4.2 (just replaced with 2.4.3 but still getting the
> > errors mentioned above in the logs. Haven't seen it run out of ports yet
> > but it has only been a few minutes)
> >Fedora release 11, kernel 2.6.30.10-105.2.23.fc11.x86_64
> >
> > pertinent opensips.cfg stuff
> > modparam("dialog", "dlg_flag", 13)
> > # db_mode: 0 - NO_DB, 1 - REALTIME, 2 - DELAYED, 3 - ON SHUTDOWN
> > modparam("dialog", "db_mode", 3)
> > modparam("mediaproxy", "mediaproxy_socket",
> > "/var/run/mediaproxy-dispatcher.sock")
> > modparam("mediaproxy", "mediaproxy_timeout", 500)
> > modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)")
> > modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)")
> > modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
> > modparam("mi_datagram", "children_count", 4)
> >
> >  if (is_method("INVITE")) {
> >  engage_media_proxy();
> >  fix_nated_sdp("8",""); # topology
> > hide the sdp o header.
> >  }
> > pertinent /etc/mediaproxy/config.ini stuff:
> > [Dispatcher]
> > socket_path = /var/run/mediaproxy-dispatcher.sock
> > [OpenSIPS]
> > socket_path = /tmp/opensips.sock
> >
>
> This looks fine.
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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[OpenSIPS-Users] opensips coredump on stop/restart

[Dave Singer]

On my production servers, both 1.6.2 and 1.6.3 versions of opensips, almost
every time I restart opensips it creates a core dump. Since I'm giving it
2GB shared mem, it takes a little while to write the core to disk and start
running again.
So a couple questions.
1. Am I giving it more memory than it needs? Judging by how much the core
dump compressed probably way to much. I'm running about 30 calls per sec
with up to 1000 active calls. Calls are using mediaproxy with engage and
thus also dialog. I did run into memory shortage problems before I
explicitly upped it to 2GB with -m 2000 on the cmd line for starting
opensips.
2. Can someone look at the core dump(s) or let me know what/how to look at
them? The 2GB core dumps compressed very small and are accessible via http:
 opensips-1.6.2-notls.core.gz
6MB
 opensips-1.6.3-tls_1.core.gz
2MB
 opensips-1.6.3-tls_2.core.gz
2MB

Thanks for any help or suggestions.
Dave
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[OpenSIPS-Users] Mediaproxy broken

[Dave Singer]

I have one server where mediaproxy is having serious problems.
In the logs I have a lot of:
   media-dispatcher[6067]: error: failed to end dialog: 500 Operation failed
and a lot of of the following but much less than the above:
   media-dispatcher[6067]: error: Got `remove' command from OpenSIPS for
unknown session with call-id `83335711-572b-122e-57ab-0024e869f3a5'

After some preiod of time, from a few minutes to a day or so, media-relay
appears to start holding open ports that have long gone dead.
After a while it runs out of ports for new calls and opensips sends the call
on without using mediaproxy.
Traffic graphs support this conclusion. They also show that it was working
fine for more than a month then stopped working.

I'm using
  opensips 1.6.2-notls (x86_64/linux)
  mediaproxy is enacted with engage_media_proxy() with dialog write to db on
shutdown.
  mediaproxy 2.4.2 (just replaced with 2.4.3 but still getting the errors
mentioned above in the logs. Haven't seen it run out of ports yet but it has
only been a few minutes)
  Fedora release 11, kernel 2.6.30.10-105.2.23.fc11.x86_64

pertinent opensips.cfg stuff
modparam("dialog", "dlg_flag", 13)
# db_mode: 0 - NO_DB, 1 - REALTIME, 2 - DELAYED, 3 - ON SHUTDOWN
modparam("dialog", "db_mode", 3)
modparam("mediaproxy", "mediaproxy_socket",
"/var/run/mediaproxy-dispatcher.sock")
modparam("mediaproxy", "mediaproxy_timeout", 500)
modparam("mediaproxy", "signaling_ip_avp", "$avp(s:nat_ip)")
modparam("mediaproxy", "media_relay_avp", "$avp(s:media_relay)")
modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
modparam("mi_datagram", "children_count", 4)

if (is_method("INVITE")) {
engage_media_proxy();
fix_nated_sdp("8",""); # topology hide
the sdp o header.
}
pertinent /etc/mediaproxy/config.ini stuff:
[Dispatcher]
socket_path = /var/run/mediaproxy-dispatcher.sock
[OpenSIPS]
socket_path = /tmp/opensips.sock



Thanks for any thoughts.

Dave
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Re: [OpenSIPS-Users] any reason why Opensips would replace 503 with 500 ?

[Dave Singer]

Thank you very much!!
I'm doing class 4 switching and for some reason all my customers expect a
503 as a slow down your sending me too much traffic. Not the I'm broken.
Is the 500 the proper way to tell them to back off a little. Is there a
header that should be used to tell them how much to back off?

Thanks again Bogdan for all your work and help you give everyone. I don't
know how you keep up.

Dave

On Fri, Oct 8, 2010 at 4:32 AM, Bogdan-Andrei Iancu
wrote:

> Hi Dave,
>
> See: http://www.opensips.org/Resources/DocsCoreFcn16#toc34  (added the
> docs for this module)
>
> Regards,
> Bogdan
>
> Dave Singer wrote:
> > Julien,
> >
> > I have been catching it in failure_route and sending it on up with this:
> > if (t_check_status("^503$")) {
> > t_reply("503", "Service Unavailable");
> > exit;
> > }
> >
> > I think you could use t_reply("503", $(rr));   (note the use of
> >  to indicate the reply context see usage examples near the top
> > of this link: <http://www.opensips.org/Resources/DocsCoreVar16> ) to
> > have it use the original reason text instead of always sending
> > "Service Unavailable" no matter what reason came with the 503. I
> > haven't tried it yet myself.
> > I'm doing this because I'm just using it as a proxy to mangle headers
> > and not decide where to go next. So I really do just want to pass the
> > response back to the server doing the routing logic. It would seem
> > passing the 503 would be the normal behavior in this case. Plus like
> > the situation of Julien which I also have, where the upstream routers
> > are not properly handling the diff between a 500 and 503.
> >
> > So I agree that it would be nice if there were a param and/or var that
> > could be set for the desired behavior like the invite timeout settings
> > for tm.
> >
> > Bogdan,
> >
> > Is "disable_503_translation" ( from the subject of the link you posted
> > ) an actual parameter (un)documented?
> >
> > On Thu, Oct 7, 2010 at 2:44 PM, thrillerbee  > <mailto:thriller...@gmail.com>> wrote:
> >
> > Bogdan,
> >
> > Is this or could this be configurable?  It, unfortunately, breaks
> > a lot of existing applications.
> >
> > Thanks.
> >
> >
> > On Thu, Oct 7, 2010 at 1:15 PM, Bogdan-Andrei Iancu
> > mailto:bog...@voice-system.ro>> wrote:
> >
> > Hi Julien,
> >
> > see:
> >
> http://lists.opensips.org/pipermail/users/2010-September/014505.html
> >
> > Regards,
> > Bogdan
> >
> > Julien Chavanton wrote:
> > > Hi, is there any reason why Opensips would replace 503 with
> > 500 ?
> > >
> > > The UA initiating the call expect 503 to reroute somewhere else
> > >
> > > INVITE :
> > >
> > >  10.0.20.14(UA) -> 10.2.0.1(Proxy) -> 10.0.4.202(UA)
> > >
> > > RESPONSE :
> > >
> > > U 10.0.4.202:5060 <http://10.0.4.202:5060> -> 10.2.0.1:5060
> > <http://10.2.0.1:5060>
> > > SIP/2.0 503 Service Unavailable.
> > >
> > > U 10.2.0.1:5060 <http://10.2.0.1:5060> -> 10.0.20.14:53584
> > <http://10.0.20.14:53584>
> > > SIP/2.0 500 Service Unavailable.
> > >
> > >
> >
> 
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> >
> >
> > --
> > Bogdan-Andrei Iancu
> > OpenSIPS Bootcamp
> > 15 - 19 November 2010, Edison, New Jersey, USA
> > www.voice-system.ro <http://www.voice-system.ro>
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] any reason why Opensips would replace 503 with 500 ?

[Dave Singer]

Julien,

I have been catching it in failure_route and sending it on up with this:
if (t_check_status("^503$")) {
t_reply("503", "Service Unavailable");
exit;
}

I think you could use t_reply("503", $(rr));   (note the use of
 to indicate the reply context see usage examples near the top of
this link:  ) to have it
use the original reason text instead of always sending "Service Unavailable"
no matter what reason came with the 503. I haven't tried it yet myself.
I'm doing this because I'm just using it as a proxy to mangle headers and
not decide where to go next. So I really do just want to pass the response
back to the server doing the routing logic. It would seem passing the 503
would be the normal behavior in this case. Plus like the situation of Julien
which I also have, where the upstream routers are not properly handling the
diff between a 500 and 503.

So I agree that it would be nice if there were a param and/or var that could
be set for the desired behavior like the invite timeout settings for tm.

Bogdan,

Is "disable_503_translation" ( from the subject of the link you posted ) an
actual parameter (un)documented?

On Thu, Oct 7, 2010 at 2:44 PM, thrillerbee  wrote:

> Bogdan,
>
> Is this or could this be configurable?  It, unfortunately, breaks a lot of
> existing applications.
>
> Thanks.
>
>
> On Thu, Oct 7, 2010 at 1:15 PM, Bogdan-Andrei Iancu <
> bog...@voice-system.ro> wrote:
>
>> Hi Julien,
>>
>> see:
>> http://lists.opensips.org/pipermail/users/2010-September/014505.html
>>
>> Regards,
>> Bogdan
>>
>> Julien Chavanton wrote:
>> > Hi, is there any reason why Opensips would replace 503 with 500 ?
>> >
>> > The UA initiating the call expect 503 to reroute somewhere else
>> >
>> > INVITE :
>> >
>> >  10.0.20.14(UA) -> 10.2.0.1(Proxy) -> 10.0.4.202(UA)
>> >
>> > RESPONSE :
>> >
>> > U 10.0.4.202:5060 -> 10.2.0.1:5060
>> > SIP/2.0 503 Service Unavailable.
>> >
>> > U 10.2.0.1:5060 -> 10.0.20.14:53584
>> > SIP/2.0 500 Service Unavailable.
>> >
>> > 
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 15 - 19 November 2010, Edison, New Jersey, USA
>> www.voice-system.ro
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] get error with opensip-cp on centos 5

[Dave Singer]

Setting setenforce 0 is only active for the current running session of the
server. The problem will be back after the reboot.
While that is fine to temporarily do that to see if SELinux is the thing
blocking, it is generally very bad to use that to solve the problem and you
definitely wouldn't want to have it like that on a production environment.
I have been getting more and more frustrated with SELinux "silently"
blocking things. I really need to take some time and understand how to
properly make use of it and work with it. Command restorecon can be a life
saver. Look up how to use it.

A good quote from another formu. It is about an apache cgi script but the
SELinux stuff is very relevant:

It is a security risk that someone could alter the script and do
something you didn't intend as root.
I would suggest *at*most* setting "SELINUX=permissive" in
/etc/selinux/config. Then you'll still have a log of all the things
that shouldn't have happened, if you ever need to look.
A better option would be extract the AVC denials from the log file and
run them through audit2allow to create a policy you could incorporate
into the running SELinux policy. Do that as often as you need to
address all the denials that are stopping your script from working.
The best option is, as others posters have suggested, figure out why
your application architecture is screwed up so much that it only works
if it runs as root, then fix it. SELinux basically fulfills two
security functions. It stops bad people from doing bad things, and it
stops foolish people from doing stupid things. Sometimes it needs
tuning, but if you just shut it off, it can't do anything for you.


On Tue, Sep 14, 2010 at 3:44 AM, ha do  wrote:

> if someone else got the same problem with me, to solve the issue please
> follow
> the link
>
> http://lists.opensips.org/pipermail/users/2010-June/013139.html
>
> Thank you
> Ha`
>
>
>
> - Original Message 
> From: ha do 
> To: OpenSIPS 
> Sent: Tue, September 14, 2010 11:12:47 AM
> Subject: [OpenSIPS-Users] get error  with opensip-cp on centos 5
>
> Hi all
>
> i try to use the opensips-cp 4:
> i login the the web page http://192.168.1.100/cp successfull
>
> the menu Admin + User are working properly
> but the system menu does not work
> i get the error on the webpage:
>
> sorry -- cannot open write fifo
>
> and this is the error in debug message :
> Sep 14 00:10:42 localhost kernel: type=1400 audit(1284437442.630:49): avc:
> denied  { getattr } for  pid=2093 comm="httpd" path="/tmp/opensips_fifo"
> dev=dm-0 ino=884744 scontext=root:system_r:httpd_t:s0
> tcontext=root:object_r:tmp_t:s0 tclass=fifo_file
>
>
> i did set fifo_mode, 0666 in opensips.cfg
>
> please help,what should i do to resolve problem
>
> Thank you
> Ha`
>
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
> ___
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>
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Re: [OpenSIPS-Users] Out of mem all of the sudden

[Dave Singer]

I ran into out of memory and adjusted the shared memory opensips uses with
the -m option.
It tells opensips how much shared memory to use in MB. eg:
.../sbin/opensips -m 500 -l eth0
gives it 500MB of shared memory and listen on eth0
I believe by default it uses 100MB if -m is not specified.

Dave

On Wed, Sep 15, 2010 at 8:20 AM, k1028  wrote:

>
> I am using opensips 1.6.3 with very only few modules loaded. The OpenSIPS
> is
> up for over a months with up any issue until now. I also tried to increase
> the PKG to 4mb and SHM to 256 mb still crash.
>
> loadmodule "db_mysql.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "rr.so"
> loadmodule "maxfwd.so"
> loadmodule "textops.so"
> loadmodule "mi_fifo.so"
> loadmodule "uri.so"
> loadmodule "dispatcher.so"
> loadmodule "mediaproxy.so"
> loadmodule "nathelper.so"
> loadmodule "dialog.so"
> loadmodule "mi_datagram.so"
> loadmodule "signaling.so"
> loadmodule "localcache.so"
> loadmodule "sst.so"
> loadmodule "avpops.so"
> loadmodule "load_balancer.so"
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Out-of-mem-all-of-the-sudden-tp5534338p5534792.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
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>
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Re: [OpenSIPS-Users] Monit error under OpenSIPS 1.6.3 and Opensips-cp 4

[Dave Singer]

In the init script, the config check where it actually runs opensips (after
running it with -c option) to check the config is very new and I just
realized and fixed that it would sometimes not kill the test opensips after
running it.
I attached my fixed version in case you are interested.

On Wed, Sep 8, 2010 at 10:42 PM, Deon Vermeulen wrote:

> Thank you very much Dave.
>
> I'll look into this and give feedback.
>
> regards
> Deon
>


opensips
Description: Binary data
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Re: [OpenSIPS-Users] Monit error under OpenSIPS 1.6.3 and Opensips-cp 4

[Dave Singer]

You may not have monit running as a service. Check the current status of
monit:
  monit status
It will tell you if it is running.
If it is not start it with:
  /etc/init.d/monit start
I primarily use centos and fedora but I think that will work with debian
systems too.
I think that the default config does not have the daemon set to listen for
communication. It needs this for commands like "monit status" and "monit
monitor opensips" and "monit unmonitor opensips"
something like this:
  set httpd port 2812 and
 use address localhost  # only accept connection from localhost
 allow localhost
Also you don't want to use the same call to the init.d script that is used
to start it up regularly otherwise you get a loop going or at least double
notify of (un)monitor actions.
What I do for the opensips portion of the monit config is have it call
/etc/init.d/opensips monit_start.
Then in /etc/init.d/opensips I have it catch that arg and not tell monit to
monitor but call the regular start function. When it gets the regular start
arg it calls the regular start function and then tells monit to start
monitoring it.
Also when the init.d script gets a restart command I don't update monit It
checks the pid file itself and realizes that the pid has changed and sends a
message to that affect rather then two messages, one that it us unmonitoring
and one that it is monitoring.
I've attached the /etc/init.d/opensips that I'm currently using.

Don't forget to make sure that the monit daemon will start on next reboot.

The attached init.d/opensips script uses some of redhats conventions for
displaying success/failure and adjust the working directory according to
your install. But I think overall it should be fairly portable to other
distros.

Note. Many of the options I pass to opensips are usually set in the script.
For the passed options to work they can not be set in opensips.cfg.
You may also notice that for a restart it validates the config before
allowing a restart. It does two levels of checks, the opensips -C to verify
the syntax of the config and actually starts a minimal opensips because
sometimes there are things that will make it not work even when the syntax
checks out.
The test run listens on lo:2056 so beware incase of conflicts. YMMV

Hope that helps.

Dave

On Wed, Sep 8, 2010 at 1:41 AM, Deon Vermeulen wrote:

> Hi Everyone
>
> I've successfully installed Opensips-cp 4 and got the Statistics monitoring
> to work, but I'm unable to get Monit to work.
>
> I've been all over google and tested recommendations, but with no luck.
>
> This is the procedure I followed to install Monit:
> *
> *
> *Step 1*: Install Monit:
> apt-get install monit
> *
> *
> *Step 2*: Once installed, you'll find the main configuration file.
> vi /etc/monit/monitrc
> set daemon  120
> set logfile syslog facility log_daemon
> set alert r...@localhost
> set httpd port 2812 and
> use address yourdomain.com
> allow localhost   # allow localhost to connect to the server and
> allow youripaddress# allow 192.168.1.2 to connect to the server,
> ###(What IP should this be? The IP of the this Box or where I am connecting
> from?)
>   # You can give only one per entry
> allow admin:monit# user name and password for
> authentication.  ###(I guess the Username and Password will be admin:admin
> if my login to Opensips-cp ios admin:admin)
> check process opensips with pidfile /var/run/opensips.pid
>
> ###(For Some Reason this does not work. Had to comment all).
> #Below is actions taken by monit when service got stuck.
> start program = "/etc/init.d/opensips start"
> stop program  = "/etc/init.d/opensips stop"
> # Admin will notify by mail if below of the condition satisfied.
> if cpu is greater than 70% for 2 cycles then alert
> if cpu > 90% for 5 cycles then restart
> *
> *
> *Step 3*: After modifying the configuration file, you should check for the
> syntax to
> make sure that everything is correct. To do this, run:
> # monit -t   (When I run this I get syntax OK)
> *
> *
> *Step 4:* Edit the /etc/default/monit file and change the parameters as
> follows:
> # You must set this variable to for 1 monit to start
> startup=1
> # To change the intervals which monit should run uncomment
> # and change this variable.
> # CHECK_INTERVALS=180
> *
> *
> *
> *
> *Configuring OpenSIPS Control Panel*
> *
> *
> *Step 1*: Configure the database access parameters. Edit the db.inc.phpfile, 
> which is
> valid for all modules. You may change the database parameters for a single
> module
> inside the module configuration section.
> For example:
> cd /var/www/opensips-cp/config
> vi db.inc.php
> //database host
> $config->db_host = "localhost";
> //database port - leave empty for default
> $config->db_port = "";
> //database connection user
> $config->db_user = "root";
> //database connection password
> $config->db_pass = "opensips";
> //database name
> $config->db_name = "opensips";
> if ($config->db_port != "")$config->db_host=$