Re: [SR-Users] [sr-dev] Planning Kamailio v5.0.1

[Brandon Armstead]

Daniel,

  +1 to what Olle said!  I have been able to produce pretty consistently a
segfault in a total dev environment evapi / uac_req but it could be
something totally crazy I'm doing, I'll get you a bit more solid info on
the exact issue in the early am here, as well as make sure that the issue
is not already resolved in latest head branch, thanks!


On Wed, Apr 5, 2017 at 1:59 AM Daniel-Constantin Mierla 
wrote:

> Hello,
>
> preparing of v5.0.1 will start very soon, if anyone needs to push
> commits to the branch 5.0 after 12:00UTC, then write first to sr-dev
> mailing list on sync on #kamailio IRC channel. Once the announcement of
> the release is out, the backports can be done again directly.
>
> Cheers,
> Daniel
>
> On 04.04.17 12:30, Daniel-Constantin Mierla wrote:
> > Hello,
> >
> > just a reminder about the plan to release v5.0.1 tomorrow. Besides
> > reporting issues, if there are some patches not backported yet, do reply
> > about them or, if a developer, cherry-pick them in branch 5.0.
> >
> > Cheers,
> > Daniel
> >
> >
> > On 28.03.17 09:40, Daniel-Constantin Mierla wrote:
> >> Hello,
> >>
> >> I am considering to release Kamailio v5.0.1 sometime next week, likely
> >> on Wednesday, April 5, 2017. Should anyone be aware of issues not listed
> >> yet on bug tracker, report them there as soon as possible to try to fix.
> >>
> >> Soon after, we should release a new version from branch 4.4 and the last
> >> one from branch 4.3.
> >>
> >> Cheers,
> >> Daniel
> >>
>
> --
> Daniel-Constantin Mierla
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
>
>
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Re: [SR-Users] Use of variables for force_send_socket

[Brandon Armstead]

Just a guess, have you tried:

modparam("pv", "varset", "externalip=s:udp:1.2.3.4:5060")
or
modparam("pv", "varset", "externalip=s:tcp:1.2.3.4:5060")

etc...

Also maybe instead of force_send_socket method, a direct assignment?

$fs = $var(externalip);



On Tue, Apr 4, 2017 at 8:14 AM, Jöran Vinzens  wrote:

> Hi all,
>
> i try to re-arrange my config and i try to use a variable declared in
> modparam pv such like:
> modparam("pv", "varset", "externalip=s:1.2.3.4")
>
> and use it as:
> force_send_socket($var(externalip));
>
> unfortunately the kamailio does not start with this configuration and
> telling me bad config file like:
> line 109, column 36: bad argument, [proto:]host[:port] expected
>
> it seems the force_send_socket is not able to get variable. Is there any
> other way to have static content defined and put across the config?
>
> Reason for the question is the possibility to use ansible as deployment
> tool. I set all the variables i need in a separate file and include it.
> This file contains the entire modconfig and global vars section to adapt
> this part to every single machine individually. This does not work if i
> need to put "constant" variables on different places in my config.
>
> If anybody has ever made this, could you please help me, or give any
> suggestions?
>
>
> Many thanks!
> BR
> Jöran
>
>
> --
>
> Jöran Vinzens - vinz...@sipgate.de
>
> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
> HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
> Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
>
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>
>
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>


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Re: [SR-Users] Several Asterisk on this same IP

[Brandon Armstead]

Not entirely sure but you may be able to range them 0-0 for dynamic
allocation and let the kernel handle it.

On Sun, Mar 12, 2017 at 12:56 PM Alex Balashov 
wrote:

> Can you not assign different Asterisk instances different ranges of RTP
> ports to allocate from?
>
> On March 12, 2017 3:47:22 PM EDT, przeqpiciel 
> wrote:
> >I would like to ask you how you deal with several asterisks and
> >kamailio on
> >that same IP address,  I have installation where i route 5060 to
> >internal
> >server with kamailio and I would like to route RTP to asterisks, but is
> >any
> >way to get around a problem with RTP ports collisions?
>
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
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Re: [SR-Users] sip over websockets sip uri

[Brandon Armstead]

Why not do UA -> kamailio 1 -> kamailio 2 or kamailio 3?

Not quite sure on what you are trying to achieve.  Think of the request uri
as the current destined address/hop if no outbound proxy host is specified
as otherwise.

On Tue, Feb 28, 2017 at 12:41 AM Jade SZ  wrote:

> Thanks for the reply. Just to be clear is the above mentioned idea
> conceptually wrong? so here is a detail of scenario
>
> All UA's are over websockets.
>
> UA --wss--> HAProxy ws-> (Identical Kamailio Servers - SIP1
> and SIP2 for now)
>
> ofcourse if a peer is registered on SIP1, SIP2 routes call to SIP1 so as
> to avoid outbound proxy or path handling. Main idea is to have it catered
> centrally.
>
> Also SIP URI domain is being used for registrations only at SIP1 and SIP2,
> however SIP1 and SIP2 have their own domain which they use in route and via
>
> So the main question is what is the significance of SIP URI in websocket
> connection? can it be dealt with above scenario?
>
>
> Regards,
> Jade
>
> On Tue, Feb 28, 2017 at 1:18 PM, Brandon Armstead 
> wrote:
>
> Perhaps you could configure an outbound proxy in the ws/wss signaling?  So
> it sends traffic to X domain/realm but goes to actual true remote proxy
> endpoint?
>
> On Mon, Feb 27, 2017 at 11:04 PM, Jade SZ  wrote:
>
> Hi Team,
>
> Question:
>
> Just wanted to clarify regarding SIP URI in webrtc (over sip) connection.
>
> e.g. if we have a scenario where Kamailio is hosted with websocket
> support. Websocket URI is used to send packets to to wss/ws address and the
> SIP URI goes with it. I have tested it with any SIP URI including any
> domain and it works if I authenticate that domain, regardless that domain
> is real or not.
>
> // client side SIP URI config
>
> var configuration = {
>   uri : 'sip:al...@example.com',
> };
>
>
> Can we use any central domain for sip uri similar to realm concept and
> send traffic to multiple kamailio servers through websocket LB.
>
> dummy.domain - for ws/wss SIP URI
>
> kamailio-a.domain.com
> kamailio-b.domain.com
>
> Thanks in advance.
>
>
> Regards,
> Jade
>
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>
>
>
>
> --
> Sincerely,
> Brandon Armstead
> CTO / CRYY.com
>
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Re: [SR-Users] sip over websockets sip uri

[Brandon Armstead]

Perhaps you could configure an outbound proxy in the ws/wss signaling?  So
it sends traffic to X domain/realm but goes to actual true remote proxy
endpoint?

On Mon, Feb 27, 2017 at 11:04 PM, Jade SZ  wrote:

> Hi Team,
>
> Question:
>
> Just wanted to clarify regarding SIP URI in webrtc (over sip) connection.
>
> e.g. if we have a scenario where Kamailio is hosted with websocket
> support. Websocket URI is used to send packets to to wss/ws address and the
> SIP URI goes with it. I have tested it with any SIP URI including any
> domain and it works if I authenticate that domain, regardless that domain
> is real or not.
>
> // client side SIP URI config
>
> var configuration = {
>   uri : 'sip:al...@example.com',
> };
>
>
> Can we use any central domain for sip uri similar to realm concept and
> send traffic to multiple kamailio servers through websocket LB.
>
> dummy.domain - for ws/wss SIP URI
>
> kamailio-a.domain.com
> kamailio-b.domain.com
>
> Thanks in advance.
>
>
> Regards,
> Jade
>
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>


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Re: [SR-Users] Merry Christmas to all Kamailians

[Brandon Armstead]

Merry Christmas!! :)
On Fri, Dec 23, 2016 at 6:45 AM Daniel Tryba  wrote:

> On Fri, Dec 23, 2016 at 01:22:42PM +0100, Olle E. Johansson wrote:
>
> > It’s been another great year for Kamailio and I’m proud to be part of
> the Kamailio development community.
>
> > We’ve made great releases, had a great conference and overall done good
> stuff :-)
>
>
>
> I have to say kamailio was very stable this year, so thank all of you
>
> who made this possible. The same for the people behind rptengine
>
> development.
>
>
>
>
>
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Re: [SR-Users] regular expression question

[Brandon Armstead]

if(!($ua =~ "")){
On Fri, Dec 9, 2016 at 7:04 AM Satish Patel  wrote:

> I am trying to block SIP scanner so i am trying to use following logic
>
> only allow "My-UserAgent" and block rest but its throwing error, but
>
> if i use  =~ regular expression which works! why negative match
>
> doesn't work?
>
>
>
> if($ua !~ "(My-UserAgent") {
>
>xlog("L_INFO","On more scriptkiddie, coming from
>
> $si, blocking");
>
>exit;
>
> }
>
>
>
>
>
> what could be wrong here?
>
>
>
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Re: [SR-Users] VERY IMPORTANT: deciding when to remove the MI code

[Brandon Armstead]

+1 - I'm all for cleaning up any technical debt and moving on with more
normalized concept.

:)

Sincerely,
Brandon Armstead

On Thu, Dec 1, 2016 at 6:17 AM Daniel-Constantin Mierla 
wrote:

> Hello,
>
> we started discussing about removing MI (so called management interface)
> for very long time, more or less since 2008. The RPC should remain the
> control interface, given its better structure for commands, parameters,
> etc ... MI is custom protocol using a line-oriented communication via
> fifo or socket file with kamailio (e.g., implemented mi_fifo and
> mi_datagram modules). RPC is the alternative, a more standardized
> concept, with better structured format.
>
> I think it's time to set a clear roadmap for doing the removal. Overall,
> it will be easier to maintain the code, right now being duplicated code
> for doing the same operation over MI or RPC, and MI shows its
> limitations (or complexity to deal with) for advanced needs (see the
> discussions about how to provide multi-line value parameters over MI).
>
> So, I want to know if there are many relying on MI directly and they
> still want to keep it, what would be the expected duration they need for
> upgrading their tools to work with RPC interface, other relevant aspects
> people have in favour of mi vs rpc.
>
> I am even willing to do the removal in time befire freezing the 5.0
> branch. We will ensure a clean start of 5.x series.
>
> The main concern from my point of view is kamctl -- but I think we can
> preserve the compatibility for kamctl commands and parameters (so
> command line execution of kamctl will be the same), but the output might
> be different. That's because it should be easy to updated it to
> communicate with jsonrpc-s module, but then it will get json-formatted
> results.
>
> To summarize, two big questions to answer:
>
> a) Are you ok to remove the MI code/commands?
>
> b) If yes to a), are you ok to be done for v5.0?
>
> Not providing feedback will be considered as 'yes' for both questions,
> so **speak up if you want MI to be kept or delay it removal**.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
>
>
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Re: [SR-Users] Identical Kamailio setup on 2 hosts

[Brandon Armstead]

Do a lookup of source ip(s) user is registered to and route appropriately
and or share user location database / replicate.  Either way however you
want original source responding to proper AOR, so just do a little
conditional checking and re route appropriately between the two.

On Sunday, September 11, 2016, Infinicalls Infinicalls <
infinica...@gmail.com> wrote:

> Hi,
>  I have two identical hosts running Kamailio with the same set of
> users. They are located in different locations. Both are running
> behind NAT and I've enabled rtpproxy for that and also advertised the
> public IP. Both the MySQL DBs are similar and have the same data.
>
> Both the hosts have different sub-domains in alias and I have set
> auto_alias=no
>
> Now, the problem is my users are not able to communicate from one
> server to another.
>
> us...@mydomain.com  from host1 is not able to communicate to
> us...@mydomain.com  at host2, though inter-server
> communication is
> happening nicely.
>
> Any idea how to solve this? Thanks.
>
> regards
> Ganesh Kumar
>
>
>
>
>
>
> --
> ---
> http://www.infinicalls.com
>
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Re: [SR-Users] Identical Kamailio setup on 2 hosts

[Brandon Armstead]

On Sunday, September 11, 2016, Infinicalls Infinicalls <
infinica...@gmail.com> wrote:

> Hi,
>  I have two identical hosts running Kamailio with the same set of
> users. They are located in different locations. Both are running
> behind NAT and I've enabled rtpproxy for that and also advertised the
> public IP. Both the MySQL DBs are similar and have the same data.
>
> Both the hosts have different sub-domains in alias and I have set
> auto_alias=no
>
> Now, the problem is my users are not able to communicate from one
> server to another.
>
> us...@mydomain.com  from host1 is not able to communicate to
> us...@mydomain.com  at host2, though inter-server
> communication is
> happening nicely.
>
> Any idea how to solve this? Thanks.
>
> regards
> Ganesh Kumar
>
>
>
>
>
>
> --
> ---
> http://www.infinicalls.com
>
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Re: [SR-Users] another bug with JSONRPC-S?

[Brandon Armstead]

Actually my apologies, just tested here on this end the correct parameter
name is "htable" not "name" as well, so you will want to adjust it as
follows to:

curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":
"2.0", "method": "htable.dump", "params" : {"htable": "callstats"}}'
http://1.2.3.4:5060/jsonrpc

Sincerely,
Brandon Armstead

On Tue, Apr 19, 2016 at 3:40 PM, Brandon Armstead  wrote:

> I believe what you want is something more along the lines of:
>
> curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":
> "2.0", "method": "htable.dump", "params" : {"name": "callstats"}}'
> http://1.2.3.4:5060/jsonrpc
>
> Sincerely,
> Brandon Armstead
>
> On Tue, Apr 19, 2016 at 3:33 PM, Brandon Armstead 
> wrote:
>
>> Brooks,
>>
>> Here is an example call:
>>
>> Ijsonrpc_exec('{"jsonrpc": "2.0", "method": "dlg.end_dlg", "params" :
>> {"h_entry": "$var(h_entry)", "h_id": "$var(h_id)"}, "reply_name": "
>> kamailio_jsonrpc_reply_fifo", "id": 1}')
>>
>> It looks like you are not specifying params correctly perhaps?  Just a
>> quick guess from glancing..
>>
>>
>> Sincerely,
>>
>> Brandon Armstead
>>
>> On Tue, Apr 19, 2016 at 10:48 AM, Brooks Bridges  wrote:
>>
>>> I may have stumbled across another bug with this module.  I believe the
>>> htable.dump method isn’t correctly recognizing that a table name is being
>>> provided:
>>>
>>>
>>>
>>> [root@server ~]# curl -s --header 'Content-Type: application/json'
>>> --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name":
>>> "callstats"}' http://1.2.3.4:5060/jsonrpc
>>>
>>> {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}}
>>>
>>>
>>>
>>> Debug log:
>>>
>>>
>>>
>>> 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]:
>>> DEBUG:  [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No
>>> htable name given#015#012Sia: SIP/2.0/TCP
>>> 1.2.3.4:54912#015#012Content-Type
>>> <http://1.2.3.4:54912#015%23012Content-Type>:
>>> application/json#015#012Server: kamailio (4.4.0-rc1
>>> (x86_64/linux))#015#012Content-Length:
>>> 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution
>>> Error"}}
>>>
>>>
>>>
>>>
>>>
>>> *Brooks Bridges | *Sr. Voice Services Engineer
>>>
>>> *O1 Communications*
>>>
>>> 5190 Golden Foothill Pkwy
>>>
>>> El Dorado Hills, CA 95762
>>>
>>> *office:* 916.235.2097 | *main:* 888.444., Option 2
>>>
>>> *email:* bbrid...@o1.com | *web:* www.o1.com
>>>
>>>
>>>
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>>>
>>
>
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Re: [SR-Users] another bug with JSONRPC-S?

[Brandon Armstead]

I believe what you want is something more along the lines of:

curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":
"2.0", "method": "htable.dump", "params" : {"name": "callstats"}}'
http://1.2.3.4:5060/jsonrpc

Sincerely,
Brandon Armstead

On Tue, Apr 19, 2016 at 3:33 PM, Brandon Armstead  wrote:

> Brooks,
>
> Here is an example call:
>
> Ijsonrpc_exec('{"jsonrpc": "2.0", "method": "dlg.end_dlg", "params" :
> {"h_entry": "$var(h_entry)", "h_id": "$var(h_id)"}, "reply_name": "
> kamailio_jsonrpc_reply_fifo", "id": 1}')
>
> It looks like you are not specifying params correctly perhaps?  Just a
> quick guess from glancing..
>
>
> Sincerely,
>
> Brandon Armstead
>
> On Tue, Apr 19, 2016 at 10:48 AM, Brooks Bridges  wrote:
>
>> I may have stumbled across another bug with this module.  I believe the
>> htable.dump method isn’t correctly recognizing that a table name is being
>> provided:
>>
>>
>>
>> [root@server ~]# curl -s --header 'Content-Type: application/json'
>> --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name":
>> "callstats"}' http://1.2.3.4:5060/jsonrpc
>>
>> {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}}
>>
>>
>>
>> Debug log:
>>
>>
>>
>> 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]:
>> DEBUG:  [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No
>> htable name given#015#012Sia: SIP/2.0/TCP
>> 1.2.3.4:54912#015#012Content-Type
>> <http://1.2.3.4:54912#015%23012Content-Type>:
>> application/json#015#012Server: kamailio (4.4.0-rc1
>> (x86_64/linux))#015#012Content-Length:
>> 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution
>> Error"}}
>>
>>
>>
>>
>>
>> *Brooks Bridges | *Sr. Voice Services Engineer
>>
>> *O1 Communications*
>>
>> 5190 Golden Foothill Pkwy
>>
>> El Dorado Hills, CA 95762
>>
>> *office:* 916.235.2097 | *main:* 888.444., Option 2
>>
>> *email:* bbrid...@o1.com | *web:* www.o1.com
>>
>>
>>
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>>
>>
>
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Re: [SR-Users] another bug with JSONRPC-S?

[Brandon Armstead]

Brooks,

Here is an example call:

Ijsonrpc_exec('{"jsonrpc": "2.0", "method": "dlg.end_dlg", "params" :
{"h_entry": "$var(h_entry)", "h_id": "$var(h_id)"}, "reply_name": "kamailio_
jsonrpc_reply_fifo", "id": 1}')

It looks like you are not specifying params correctly perhaps?  Just a
quick guess from glancing..


Sincerely,

Brandon Armstead

On Tue, Apr 19, 2016 at 10:48 AM, Brooks Bridges  wrote:

> I may have stumbled across another bug with this module.  I believe the
> htable.dump method isn’t correctly recognizing that a table name is being
> provided:
>
>
>
> [root@server ~]# curl -s --header 'Content-Type: application/json'
> --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name":
> "callstats"}' http://1.2.3.4:5060/jsonrpc
>
> {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}}
>
>
>
> Debug log:
>
>
>
> 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]:
> DEBUG:  [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No
> htable name given#015#012Sia: SIP/2.0/TCP
> 1.2.3.4:54912#015#012Content-Type
> <http://1.2.3.4:54912#015%23012Content-Type>:
> application/json#015#012Server: kamailio (4.4.0-rc1
> (x86_64/linux))#015#012Content-Length:
> 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution
> Error"}}
>
>
>
>
>
> *Brooks Bridges | *Sr. Voice Services Engineer
>
> *O1 Communications*
>
> 5190 Golden Foothill Pkwy
>
> El Dorado Hills, CA 95762
>
> *office:* 916.235.2097 | *main:* 888.444., Option 2
>
> *email:* bbrid...@o1.com | *web:* www.o1.com
>
>
>
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Re: [SR-Users] [Kamailio-Business] Merry Christmas and Happy Holidays!

[Brandon Armstead]

Happy Holidays!   So crazy to think how long Kamailio has been in the works and 
all of the great people I've met because of it.  Hope everyone has had a great 
2015, looking forward to an amazing 2016!

Sent from my iPhone

> On Dec 24, 2015, at 12:38 PM, Daniel-Constantin Mierla  
> wrote:
> 
> With an amazing 2015 almost gone, I am using this moment to give my
> thanks and greetings to the people involved in Kamailio project, old and
> new friends, developers, contributors, the engaged and warm community
> members.
> 
> Very soon 2016 will arrive, Kamailio will celebrate 15 years of
> development, therefore we are looking to a special year ahead!
> 
> Merry Christmas and Happy Winter Holidays!
> 
> Daniel
> 
> -- 
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> 
> 
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Re: [SR-Users] Best practise for limiting concurrent calls across a cluster

[Brandon Armstead]

I would use some kind of RESTFUL service implementation, various transport 
mediums are avail i.e. RPC etc.  What database are you currently using ?

Sincerely,
Brandon Armstead

> On Sep 3, 2015, at 5:19 PM, jay binks  wrote:
> 
> Hey all,
> 
> So I have a cluster of Kamailio servers ( 4 servers currently, soon to be 8 ),
> I'm looking for suggestions about the BEST way to achieve concurrent call 
> limiting on a per customer basis, across the whole cluster.
> 
> Initially I mis-read the dialog module documentation and assumed that dialog 
> would provide me this ability, when used with a database.  however it seems 
> that the dialog module does not pull data from the DB after the initial 
> startup.
> 
> I know I can use sql ops to increment and decrement using 
> event_route[dialog:start] and event_route[dialog:end]. however the database 
> I've chosen ( for other valid reasons ) does not have an atomic increment and 
> decrement.   I could add yet another DB, but that just adds more failure 
> points.
> 
> so lets forget my setup,  Im wanting suggestions about the BEST setup for 
> this sort of thing.   while remaining fault tolerant, and preferably without 
> relying on any single point of failure.
> 
> 
> Sincerely
> 
> Jay
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Re: [SR-Users] What could cause an intermittent fault (28) with http_query()?

[Brandon Armstead]

What happens if you run the daemon in foreground / high verbosity ?  Paste 
error logs back here.  My guess would be there is some kind of blocking 
happening, ie DNS lookup possibly try using an alternate resolver ?

Sent from my iPhone

> On Jan 30, 2015, at 10:19 AM, Tim Chubb  wrote:
> 
> Hi All
>  
> I seem to be experiencing an intermittent fault with the utils http_query() 
> method.  We are implementing a routing and white list component that is 
> accessed via a REST api, however i have observed several occasions where this 
> is logged:
> Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils 
> [functions.c:149]: http_query(): failed to perform curl (28)
>  
> The indicated error number (28) seems to suggest a timeout is occurring with 
> curl, however examining a capture of network traffic when this happens shows 
> that a http request is not sent from the server to the destination at all, 
> usually attempting a second call results in everything working correctly.
>  
> As stated its intermittent in its nature and so I cannot reliably trigger 
> this issue, other than through sheer repetition, so any ideas as to what 
> could be causing this issue would be gratefully received,
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Re: [SR-Users] Async module taking down our server

[Brandon Armstead]

Why not just kill the call and have billing fix up for minimum duration occur 
during CDR creation?  Does not make sense to delay Hangup just to meet minimum 
duration.

Sent from my iPhone

> On Jan 28, 2015, at 5:37 PM, Will Ferrer  wrote:
> 
> Hi Daniel
> 
> Yeah I am happy to be able to report the success. Thanks for everything as 
> always!
> 
> I hope you are well.
> 
> Will
> 
>> On Wed, Jan 28, 2015 at 5:54 AM, Daniel-Constantin Mierla 
>>  wrote:
>> Hello,
>> 
>> great that it was sorted out and it was not on Kamailio side :-)
>> 
>> Also, glad to hear that async processing did increase capacity to handle 
>> more concurrent calls, even it was causing troubles to other applications ...
>> 
>> Cheers,
>> Daniel
>> 
>> 
>>> On 28/01/15 05:40, Will Ferrer wrote:
>>> Hello
>>> 
>>> I wanted to give an update on this.
>>> 
>>> My business partner that found the issue and has been monitoring the 
>>> problem has tracked down the issue. It turns out that the features we 
>>> implemented using the async module were leading to more calls going on con 
>>> currently (as they were intended to) and this was causing and issue with 
>>> voip monitor. So the issue was not with the Async module.
>>> 
>>> All the best.
>>> 
>>> Will Ferrer
>>> 
>>> Switchsoft
>>> 
>>> On Mon, Jan 19, 2015 at 8:43 PM, Will Ferrer  
>>> wrote:
 Hi All
 
 We are trying to use the async module to to delay sending a bye on from 
 one end of the call to the other.
 
 We are using async_route(routename, seconds) to delay the WITHINDLG route. 
 The idea is that in the future we want to be able to have our billing min 
 duration enforced (though currently we are having issues with the dialog 
 module that we are discussing in another thread).
 
 After running this on our deploy servers, the delays before sending on the 
 byes get longer and longer, and then kamailio goes down. Then the receive 
 udp buffer fills up.
 
 We tried it with both 4 and 400 async workers, and it made no difference.
 
 I am including a screen capture of the servers stats when this happens 
 taken from voip monitor.
 
 Here are the relevant parts of the config:
 
 ...
 loadmodule "async.so"
 ...
 modparam("async", "workers", ASYNC_THREADS)
 ...
 request_route {
 ...
 route(DELAYED_BYE_STATIC);
 ...
 route[DELAYED_BYE_STATIC] {
  xlog("L_DEBUG","route DELAYED_BYE_STATIC");
  #!ifdef WITH_DELAYED_BYE_STATIC
  if (is_method("BYE")) {
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, from self \n");
  #if (from_uri == myself) {
  if ((allow_trusted() || allow_source_address()) && from_uri == myself) {
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, Bye detected, from self \n");
  send_reply("200", "OK");
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, sent 200 about to sleep \n");
  setflag(FLT_ACC); # do accounting ...
  setflag(FLT_ACCFAILED); # ... even if the transaction fails
  if (has_totag()) {
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG_DELAYED 
 \n");
  async_route("WITHINDLG_DELAYED", MIN_DURATION);
  } else {
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, sleeping to WITHINDLG \n");
  async_route("WITHINDLG", MIN_DURATION);
  }
  xlog("L_DEBUG","route DELAYED_BYE_STATIC, slept\n");
  exit;
  }
  }
  #!endif
  return;
 }
 ...
 route[WITHINDLG_DELAYED] {
  xlog("L_DEBUG", "route WITHINDLG_DELAYED: triggered \n");
  $avp(was_delayed) = 1;
  route(WITHINDLG);
 }
 ...
 route[WITHINDLG] {
  xlog("L_DEBUG", "route WITHINDLG: will -- DLG triggered, request method: 
 $rm \n");
  #!ifdef WITH_DISPATCHER
  if(is_method("BYE|CANCEL")) {
  xlog("L_DEBUG","route WITHINDLG:  cancel or bye detected, request method: 
 $rm \n");
  #!ifdef WITH_DISPATCHER_LOAD_AWARE
  xlog("L_DEBUG","route WITHINDLG: running ds_load_update, request method: 
 $rm \n");
  ds_load_update();
  #dlg_get ("$ci","$ft","$tt"); 
   #dlg_bye ("all");
  #!endif
  }
  #!endif
 
  if (has_totag() || $avp(was_delayed) == 1) {
  xlog("L_DEBUG", "route WITHINDLG: will -- DLG has totag or was_delayed: 
 $avp(was_delayed)  \n");
  # sequential request withing a dialog should
  # take the path determined by record-routing
  if (loose_route()) {
  xlog("L_DEBUG", "route WITHINDLG: will -- DLG has loose route \n");
  route(DLGURI);
  if (is_method("BYE")) {
  xlog("L_DEBUG","route WITHINDLG: BYE detected");
  setflag(FLT_ACC); # do accounting ...
  setflag(FLT_ACCFAILED); # ... even if the transaction fails
  xlog("L_DEBUG","route WITHINDLG: ACC flag set");
  
  }
  else if ( is_method("ACK") ) {
  # ACK is forwarded statelessy
  route(NATMANAGE);
  }
  else if ( is_method("NOTIFY") ) {
>>

Re: [SR-Users] Happy New Year!

[Brandon Armstead]

Happy New Year!!!

Many great new things to come.  #2015 here we are :).

- Brandon

> On Dec 31, 2014, at 2:33 PM, Mahmoud Ramadan Ali 
>  wrote:
> 
> Happy new year to Kamailio and the developers and all the mailing list users 
> and all the world !
> 
>> On Wed, Dec 31, 2014 at 11:10 PM, Daniel-Constantin Mierla 
>>  wrote:
>> A very dynamic 2014 for Kamailio has reached its end! Thank you everyone
>> for contributing to it!
>> 
>> Looking forward to 2015, a lot of new features in Kamailio and new
>> contributors! I wish a healthy and prosperous year to all Kamailio
>> friends, hoping to meet many of you at Kamailio World Conference and
>> other events around the globe!
>> 
>> Happy New Year!
>> Daniel
>> 
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> 
>> 
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Re: [SR-Users] no fork mode and more than one listen address found (will use only the first one)

[Brandon Armstead]

kamailio -D no -l eth0 -l eth1 -l eth2

Try this?


On Tue, Aug 5, 2014 at 12:21 PM, Yuriy Gorlichenko 
wrote:

> I Already use it.
> As I see my problem about to run the main kamailio process in foreground
> and fork child processes as usual.
>
>
> 2014-08-05 23:18 GMT+04:00 Brandon Armstead :
>
> Yuriy,
>>
>> You can use the listen directive multiple times.
>>
>> listen=eth0
>> listen=eth1
>>
>> etc...
>>
>>
>> On Tue, Aug 5, 2014 at 12:09 PM, Yuriy Gorlichenko 
>> wrote:
>>
>>> Hello. I Installed kamailio 4.1.4, when I starting my server I see
>>> following Warning
>>>
>>> no fork mode and more than one listen address found (will use only the
>>> first one)
>>>
>>> So I see only first listening port at my netstat
>>>
>>> At my kamailio.cfg file I write fork=yes? but it not helps me.
>>>
>>> I found the same issue at this mailing list
>>> http://comments.gmane.org/gmane.comp.voip.openser.user/10686
>>>
>>> But it very old issue (2007 year) and I think patch at this issue does
>>> not help me.
>>>
>>> How I can start my Server for runnings with many interfaces?
>>>
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>>
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Re: [SR-Users] no fork mode and more than one listen address found (will use only the first one)

[Brandon Armstead]

Yuriy,

You can use the listen directive multiple times.

listen=eth0
listen=eth1

etc...


On Tue, Aug 5, 2014 at 12:09 PM, Yuriy Gorlichenko 
wrote:

> Hello. I Installed kamailio 4.1.4, when I starting my server I see
> following Warning
>
> no fork mode and more than one listen address found (will use only the
> first one)
>
> So I see only first listening port at my netstat
>
> At my kamailio.cfg file I write fork=yes? but it not helps me.
>
> I found the same issue at this mailing list
> http://comments.gmane.org/gmane.comp.voip.openser.user/10686
>
> But it very old issue (2007 year) and I think patch at this issue does not
> help me.
>
> How I can start my Server for runnings with many interfaces?
>
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Re: [SR-Users] [Kamailio-4.1.4]Make Install Error

[Brandon Armstead]

Daniel,

I can confirm this bug on Debian.  I hasn't bothered looking into it 
further since it was simple enough to manually complete remainder of install.  
I'll pull more debug data for you when I make it into the office shortly if 
someone doesn't get to it before me.

Sent from my iPhone

> On Aug 5, 2014, at 5:54 AM, Daniel-Constantin Mierla  
> wrote:
> 
> It is working ok with bash here and this is probably the most used shell.
> 
> Are you sure you haven't changed the Makefile somehow?
> 
> Can you try running make with verbose/debugging parameter and see if there is 
> a hint in the output?
> 
> Daniel
> 
>> On 05/08/14 14:36, Djamel Bahamid wrote:
>> Hello, 
>> 
>> When I execute the command echo $SHELL, I have the following result 
>> : 
>> 
>> /bin/bash 
>> 
>> What's to be done now to avoid this error ? 
>> 
>> Best Regards, 
>> 
>> Djamel. 
>> 
>> Le 01/08/2014 22:13, Daniel-Constantin Mierla a écrit :
>>> Hello,
>>> 
>>> what is your shell? See where /sbin/sh is pointing to:
>>> 
>>> ls -l /sbin/sh
>>> 
>>> also, do:
>>> 
>>> echo $SHELL
>>> 
>>> Cheers,
>>> Daniel
>>> 
 On 01/08/14 18:20, Djamel Bahamid wrote:
 Hello everybody,
 
 I am new on this environment ( Kamailio). I try to install(settle) 
 Kamailio 4.1.4, on an O.S Suse Sles11 SP3, all the compilation took place, 
 with the exception of the installation (make install), where I obtain the 
 following error: 
 
 
 make[2]: Nothing to be done for `install-if-newer'. 
 touch   /usr/local/lib64/kamailio/modules/userblacklist.so 
 install -m 755  userblacklist.so  /usr/local/lib64/kamailio/modules 
 make[2]: `libkmi.so.1.0' is up to date. 
 make[2]: `libsrdb1.so.1.0' is up to date. 
 make[2]: `libkcore.so.1.0' is up to date. 
 make[2]: `libsrutils.so.1.0' is up to date. 
 LD (gcc) [M usrloc.so]  usrloc.so 
 make[2]: Nothing to be done for `install-if-newer'. 
 make[2]: Nothing to be done for `install-if-newer'. 
 make[2]: Nothing to be done for `install-if-newer'. 
 make[2]: Nothing to be done for `install-if-newer'. 
 touch   /usr/local/lib64/kamailio/modules/usrloc.so 
 install -m 755  usrloc.so  /usr/local/lib64/kamailio/modules 
 touch   /usr/local/lib64/kamailio/modules/xhttp.so 
 install -m 755  xhttp.so  /usr/local/lib64/kamailio/modules 
 touch   /usr/local/lib64/kamailio/modules/xhttp_rpc.so 
 install -m 755  xhttp_rpc.so  /usr/local/lib64/kamailio/modules 
 touch   /usr/local/lib64/kamailio/modules/xlog.so 
 install -m 755  xlog.so  /usr/local/lib64/kamailio/modules 
 touch   /usr/local/lib64/kamailio/modules/xprint.so 
 install -m 755  xprint.so  /usr/local/lib64/kamailio/modules 
 /bin/sh: -c: line 18: syntax error near unexpected token `fi' 
 /bin/sh: -c: line 18: ` fi ; \' 
 make: *** [install-cfg] Error 2 
 
 
 Could you help me please to solve this error,
 
 Best Regards,
 
 Djamel.
 
 
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>>> 
>>> -- 
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>>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>> 
>>> 
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Re: [SR-Users] Condition Statements - How many levels

[Brandon Armstead]

Joel,

You are looking for something like this logically:

if(!($var(is_a_present) == $null && $var(is_b_present))){
# a and b exist
} else if(!($var(is_a_present) == $null){
# a exists
} else if(!($var(is_b_present) == $null)){
# b exists
} else {
 # failover
}

Sincerely,
Brandon Armstead


On Wed, Jul 16, 2014 at 5:46 AM, Joel White  wrote:

> I was wondering how many levels deep in conditional _if_  _else_
> statements can go.  I need to create a statement that has 4 possible
> outcomes and need to know if this is possible.
>
> I have two variables
>
> I want one thing to happen if only the first variable is present
>
> Another if only the second variable is present
>
> Yet another if they are both present
>
> And a fail over if None of the variables are present
>
>
>
> I do see that is can be done with 3 conditions using   _if_  _if else_
> _else_
>
> how can I accomplish 4 conditional statements?
>
>
> Thank you in advance
>
> Joel
>
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Re: [SR-Users] Condition Statements - How many levels

[Brandon Armstead]

Joel,

   Forgot second half of initial if

if(!($var(is_a_present) == $null && $var(is_b_present) *== $null)*){
# a and b exist
} else if(!($var(is_a_present) == $null){
# a exists
} else if(!($var(is_b_present) == $null)){
# b exists
} else {
 # failover
}



On Wed, Jul 16, 2014 at 1:37 PM, Brandon Armstead  wrote:

> Joel,
>
> You are looking for something like this logically:
>
> if(!($var(is_a_present) == $null && $var(is_b_present))){
> # a and b exist
> } else if(!($var(is_a_present) == $null){
> # a exists
> } else if(!($var(is_b_present) == $null)){
> # b exists
> } else {
>  # failover
> }
>
> Sincerely,
> Brandon Armstead
>
>
> On Wed, Jul 16, 2014 at 5:46 AM, Joel White  wrote:
>
>> I was wondering how many levels deep in conditional _if_  _else_
>> statements can go.  I need to create a statement that has 4 possible
>> outcomes and need to know if this is possible.
>>
>> I have two variables
>>
>> I want one thing to happen if only the first variable is present
>>
>> Another if only the second variable is present
>>
>> Yet another if they are both present
>>
>> And a fail over if None of the variables are present
>>
>>
>>
>> I do see that is can be done with 3 conditions using   _if_  _if else_
>> _else_
>>
>> how can I accomplish 4 conditional statements?
>>
>>
>> Thank you in advance
>>
>> Joel
>>
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>>
>>
>
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Re: [SR-Users] Branch feedback behaviour

[Brandon Armstead]

Alex,

   I am not sure of your use case - but one common example I can think of
is a LRN service provided via 302 redirect.  What we do in this instance is
UAC -> Kamailio on this initial transaction we send the call directly to
the LRN service (not appending) any other branches and alter the original
state of ruri, we then set a special reply route that receives the LRN
information, caches it in a database so subsequent requests entirely avoid
this process for a period of time since its already cached.  We also set a
custom failure route that appends the new branch(s) upstream back to LCR /
PSTN using LRN information.  Now with this said - while the transaction is
in flight upstream, and a cancel from UAC is sent, simply flag the
transaction / response you receive back from upstream and drop it in the
reply.

Sincerely,
Brandon Armstead


On Sun, Jun 8, 2014 at 12:22 PM, Alex Balashov 
wrote:

> Sounds like t_drop_replies() might be the ticket, but I wanted to check
> with the Best Practices Council.
>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> Tel: +1-678-954-0670
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
> Please be kind to the English language:
>
> http://www.entrepreneur.com/article/232906
>
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Re: [SR-Users] error handling

[Brandon Armstead]

Acc_db_request

Sent from my iPhone

> On Nov 24, 2013, at 11:20 AM, Oliver Roth  wrote:
> 
> Hi all
>  
> Question about error handling with kamailio.
>  
> We send call to carrier and get back error 404.
> In carrierfailureroute we catch up this error and send call to an internal 
> freeswitch that plays a voiceprompt saying: “destination not available”
>  
> In accounting this calls is collected like a “normal” call – cause the 
> internal freeswitch did the connection.
> I would like to play the voiceprompt but get the error 404 and see the call 
> in the missed calls acc table.
>  
> With 486 (busy) it is simple because we do not need an rtp response … just 
> fast busy.
>  
> How can we handle this for errors we need to play a voiceprompt?
>  
> What we do in failure route:
>  
> failure_route[MANAGE_FAILURE] {
>  
> sip_trace();
> setflag(22);
> if (t_grep_status("486")){
> xlog("L_INFO", "Status 486 - busy");
> t_reply("486", "Busy");
> exit;
> }
>  
>  
> revert_uri();
> route(NATMANAGE);
>  
> if (t_is_canceled()) {
> exit;
> }
>  
> xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
> trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
> $avp(s:todirection) fu: $fu\n");
>  
> if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
> "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
> xlog("cr_next_domain failed");
> exit;
> }
>  
> if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
> "call_id", "$avp(s:todirection)" )){
> xlog("cr_route failed");
> exit;
> }
>  
>  
> xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
> trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
> $avp(s:todirection) fu: $fu \n");
> $avp(s:trunk_out) = $avp(s:todirection);
>  
> xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via 
> $avp(s:trunk_out) rU $rU 
>  ");
> if (is_method("INVITE"))
> {
> setflag(FLT_ACC); # do accounting
> setflag(FLT_ACCMISSED); # oro 28.10.13
> }
>  
>  
> route(ALTERHEADER);
>  
> t_on_failure("MANAGE_FAILURE");
> #append_branch();
>  
> if (!t_relay()) {
> xlog("failureroute t_relay failed");
> exit;
> }
>  
> }
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Re: [SR-Users] error handling

[Brandon Armstead]

Flag the call as missed internally in your accounting or write to the missed 
acc table there is a method in acc documentation.

Sent from my iPhone

> On Nov 24, 2013, at 11:20 AM, Oliver Roth  wrote:
> 
> Hi all
>  
> Question about error handling with kamailio.
>  
> We send call to carrier and get back error 404.
> In carrierfailureroute we catch up this error and send call to an internal 
> freeswitch that plays a voiceprompt saying: “destination not available”
>  
> In accounting this calls is collected like a “normal” call – cause the 
> internal freeswitch did the connection.
> I would like to play the voiceprompt but get the error 404 and see the call 
> in the missed calls acc table.
>  
> With 486 (busy) it is simple because we do not need an rtp response … just 
> fast busy.
>  
> How can we handle this for errors we need to play a voiceprompt?
>  
> What we do in failure route:
>  
> failure_route[MANAGE_FAILURE] {
>  
> sip_trace();
> setflag(22);
> if (t_grep_status("486")){
> xlog("L_INFO", "Status 486 - busy");
> t_reply("486", "Busy");
> exit;
> }
>  
>  
> revert_uri();
> route(NATMANAGE);
>  
> if (t_is_canceled()) {
> exit;
> }
>  
> xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
> trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
> $avp(s:todirection) fu: $fu\n");
>  
> if(!cr_next_domain("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", 
> "$avp(s:trunk_in)", "$T_reply_code", "$avp(s:tree)" )){
> xlog("cr_next_domain failed");
> exit;
> }
>  
> if(!cr_route("$avp(s:cr_pref_carr)", "$avp(s:tree)", "$rU", "$rU", 
> "call_id", "$avp(s:todirection)" )){
> xlog("cr_route failed");
> exit;
> }
>  
>  
> xlog("L_INFO", "failure_route rd: $rd replCode:  $T_reply_code 
> trunk_in: $avp(s:trunk_in) Tree: $avp(s:tree), rU:  $rU, todirection 
> $avp(s:todirection) fu: $fu \n");
> $avp(s:trunk_out) = $avp(s:todirection);
>  
> xlog(, "L_INFO", "RELAY - FailureRoute: Outbound sent via 
> $avp(s:trunk_out) rU $rU 
>  ");
> if (is_method("INVITE"))
> {
> setflag(FLT_ACC); # do accounting
> setflag(FLT_ACCMISSED); # oro 28.10.13
> }
>  
>  
> route(ALTERHEADER);
>  
> t_on_failure("MANAGE_FAILURE");
> #append_branch();
>  
> if (!t_relay()) {
> xlog("failureroute t_relay failed");
> exit;
> }
>  
> }
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[SR-Users] Kamailio 3.2 -> 3.3.5 Migration

[Brandon Armstead]

Hello,

   Just wanted to shoot a quick email to the user list and see if anyone
had any 'gotchas' to keep in mind when migrating from 3.2 to 3.3.5.

I know table structure has changed and will need to be updated, but are
there otherwise any syntax and or major configuration differences?

Thank you for your time in advance.

Sincerely,
Brandon Armstead
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Re: [SR-Users] Kamailio Admin Book

[Brandon Armstead]

Looking good - count me in for a copy :).


On Wed, Aug 7, 2013 at 11:34 AM, Daniel-Constantin Mierla  wrote:

> Hello,
>
> being asked for quite a while about this topic, I can give now more
> details about if there is or is going to be any time soon Kamailio book. We
> are just about to complete the 22nd chapter (over 280 A4 pages of content),
> with 3-5 still planned to get in this edition. More details about the
> content and structure can be found at:
>
>   * http://asipto.com/u/kab
>
> For 'early adopters', the book might be available as soon as mid of
> August, with the content at that time. They will get of course updated
> versions until the book is completely finished. We hope to use a group of
> such people for reviewing and feedback. I will follow up soon with more
> information regarding pre-release selling.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/**miconda
>
>
> __**_
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>
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Re: [SR-Users] log reply message number

[Brandon Armstead]

Are you firing t_on_reply?  Is it being called?

Sent from my iPhone

On May 15, 2013, at 11:04 AM, Robert R  wrote:

> Hi,
> 
> I am trying to add a log when receiving 1xx or 2xx reply message:
> 
> # manage incoming replies
> onreply_route[MANAGE_REPLY] {
>xdbg("incoming reply\n");
>if(status=~"[12][0-9][0-9]")
>{
>   xlog("L_INFO","reply message: $T_reply_code   ci:$ci"); 
>   route(NATMANAGE);
>}
> }
> 
> 
> Adding the above line does not log the reply message number (180 or 183 or 
> 200, ...).
> 
> Thanks,
> R
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Re: [SR-Users] How to correctly check if the variable is set?

[Brandon Armstead]

I believe what you are looking for is $null

if($avp(s:test) == $null){

}


On Mon, Jan 14, 2013 at 3:59 AM, Mino Haluz  wrote:

> Hi,
>
> how should I check if the value is set?
>
> if ($avp(s:test) == "") {
>
> or is there any null keyword ? If so, does it work for $avp, $sht, $var
> and $shv ?
>
> Thanks,
>
> Mino
>
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>
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Re: [SR-Users] [Kamailio-Business] Merry Christmas!

[Brandon Armstead]

Merry Christmas!!!


On Mon, Dec 24, 2012 at 1:00 PM, Daniel-Constantin Mierla  wrote:

> Another year getting to its end! Looking back, looks like one with the
> greatest achievements in the development of the project so far. Year is not
> done, so I save that summary for one week later, there is still stuff on
> its way to our source code repository.
>
> Now I want to thank to everyone promoting and contributing to the project,
> from developers to community members, and wish Merry Christmas and great
> winter holidays to all supporters of our project!
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
> __**_
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> http://lists.kamailio.org/cgi-**bin/mailman/listinfo/business
> http://lists.openser-project.**org/cgi-bin/mailman/listinfo/**business
>
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Re: [SR-Users] How can I directly access the domain of additional branches??

[Brandon Armstead]

Steve,

   You would do the required logic inside of branch_route.

Sincerely,
Brandon Armstead

On Fri, Oct 5, 2012 at 9:45 AM, Stephen Dodge (Bistech) <
special.proje...@bistech.co.uk> wrote:

>  Hello,
>
>
>
> Could someone advise if it is possible to directly access the domain of
> additional branches as $rd references the domain of the main URI only.
>
>
>
> If it’s not possible does anyone have any tips on how I can achieve this.
>
>
>
> Many Thanks.
>
>
>
> *Steve.*
>
>
>
> --
>
> Information in this message, including any attachments, is confidential to
> the person to whom it is addressed and may be legally privileged. If you
> are not the intended recipient please notify the sender and delete the
> message from your system. Please note that Bistech Group plc, Bistech plc,
> Bisnet Limited and the sender do not accept any responsibility for viruses.
> It is your responsibility to check the e-mail and any attachments for
> viruses. Calls may be monitored and recorded.
>
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Re: [SR-Users] [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks

[Brandon Armstead]

Samy,

  I am utilizing these functions without any success.

Essentially the problem I am running into is the SDP is correctly mangled
and negotiated, however rtpproxy simply never bridges the public side.

UAC 1 -> KAM -> UAC 2

I see RTP packets flow from UAC 1 -> KAM -> UAC 2

I also see RTP packets from from UAC 2 -> KAM

The caller does not hear any audio, as KAM/RTPProxy is not sending audio
back to the caller.

Sincerely,
Brandon Armstead

On Fri, Sep 14, 2012 at 11:12 PM, SamyGo  wrote:

> ok ignore the name in my previous email: its
> http://www.kamailio.org/docs/modules/3.3.x/modules/rtpproxy.html#id2550699
>
>
>
> On Sat, Sep 15, 2012 at 10:05 AM, Brandon Armstead wrote:
>
>> Samy,
>>
>> I am not locating any such functions, engage_rtpproxy ?
>>
>> Perhaps you could point me further into the right direction?
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> On Fri, Sep 14, 2012 at 9:13 PM, SamyGo  wrote:
>>
>>> Yes, you should find the function engage_rtpproxy on module docs and use
>>> it. It will work exactly like your old force-rtp-proxy but with more
>>> enhanced way.
>>> On Sep 15, 2012 7:34 AM, "Brandon Armstead"  wrote:
>>>
>>>> Klaus,
>>>>
>>>>It seems that force_rtp_proxy is removed in Kamailio 3.3 --- has
>>>> there been a work-around for this bridge issue without using
>>>> force_rtp_proxy?
>>>>
>>>> Sincerely,
>>>> Brandon Armstead
>>>>
>>>> On Wed, Nov 4, 2009 at 5:46 AM, Klaus Darilion <
>>>> klaus.mailingli...@pernau.at> wrote:
>>>>
>>>>>
>>>>>
>>>>> Alex Balashov schrieb:
>>>>>
>>>>>  That's pretty much what I did.
>>>>>>
>>>>>> All I had to do was use force_rtp_proxy().  Everything was broken
>>>>>> when I tried to use rtpproxy_offer/answer, though the code suggests they
>>>>>> are just wrappers around force_rtp_proxy().
>>>>>>
>>>>>
>>>>>
>>>>> That's strange. I took a look at the code and these are really just
>>>>> wrappers  - fmro code point of view it looks fine.
>>>>>
>>>>> Do you have syslog traces?
>>>>>
>>>>> regards
>>>>> klaus
>>>>>
>>>>>
>>>>>
>>>>> __**_
>>>>> Kamailio (OpenSER) - Users mailing list
>>>>> us...@lists.kamailio.org
>>>>> http://lists.kamailio.org/cgi-**bin/mailman/listinfo/users<http://lists.kamailio.org/cgi-bin/mailman/listinfo/users>
>>>>> http://lists.openser-project.**org/cgi-bin/mailman/listinfo/**users<http://lists.openser-project.org/cgi-bin/mailman/listinfo/users>
>>>>>
>>>>
>>>>
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>>>>
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>>>
>>
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>>
>
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Re: [SR-Users] [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks

[Brandon Armstead]

Samy,

I am not locating any such functions, engage_rtpproxy ?

Perhaps you could point me further into the right direction?

Sincerely,
Brandon Armstead

On Fri, Sep 14, 2012 at 9:13 PM, SamyGo  wrote:

> Yes, you should find the function engage_rtpproxy on module docs and use
> it. It will work exactly like your old force-rtp-proxy but with more
> enhanced way.
> On Sep 15, 2012 7:34 AM, "Brandon Armstead"  wrote:
>
>> Klaus,
>>
>>It seems that force_rtp_proxy is removed in Kamailio 3.3 --- has there
>> been a work-around for this bridge issue without using force_rtp_proxy?
>>
>> Sincerely,
>> Brandon Armstead
>>
>> On Wed, Nov 4, 2009 at 5:46 AM, Klaus Darilion <
>> klaus.mailingli...@pernau.at> wrote:
>>
>>>
>>>
>>> Alex Balashov schrieb:
>>>
>>>  That's pretty much what I did.
>>>>
>>>> All I had to do was use force_rtp_proxy().  Everything was broken when
>>>> I tried to use rtpproxy_offer/answer, though the code suggests they are
>>>> just wrappers around force_rtp_proxy().
>>>>
>>>
>>>
>>> That's strange. I took a look at the code and these are really just
>>> wrappers  - fmro code point of view it looks fine.
>>>
>>> Do you have syslog traces?
>>>
>>> regards
>>> klaus
>>>
>>>
>>>
>>> __**_
>>> Kamailio (OpenSER) - Users mailing list
>>> us...@lists.kamailio.org
>>> http://lists.kamailio.org/cgi-**bin/mailman/listinfo/users<http://lists.kamailio.org/cgi-bin/mailman/listinfo/users>
>>> http://lists.openser-project.**org/cgi-bin/mailman/listinfo/**users<http://lists.openser-project.org/cgi-bin/mailman/listinfo/users>
>>>
>>
>>
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Re: [SR-Users] [Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks

[Brandon Armstead]

Klaus,

   It seems that force_rtp_proxy is removed in Kamailio 3.3 --- has there
been a work-around for this bridge issue without using force_rtp_proxy?

Sincerely,
Brandon Armstead

On Wed, Nov 4, 2009 at 5:46 AM, Klaus Darilion  wrote:

>
>
> Alex Balashov schrieb:
>
>  That's pretty much what I did.
>>
>> All I had to do was use force_rtp_proxy().  Everything was broken when I
>> tried to use rtpproxy_offer/answer, though the code suggests they are just
>> wrappers around force_rtp_proxy().
>>
>
>
> That's strange. I took a look at the code and these are really just
> wrappers  - fmro code point of view it looks fine.
>
> Do you have syslog traces?
>
> regards
> klaus
>
>
>
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Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Daniel,

   One question I do have - is the $timef(%Y%m%d)  evaluated only once at
run time (when kamailio is started) - or is it evaluated at the time of the
ACC'ing action.

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 1:22 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> the $timef(...) is evaluated, but you don't have time specifiers there,
> only static letters. You have to use % in front of the letters, as I guess
> from the example:
>
> http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables
>
> Try acc_$timef(%Y%m%d).
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 10:17 PM, Brandon Armstead wrote:
>
> Daniel,
>
>  My apologies - forgot to make clean.  However still no cigar.
>
>  Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
> [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
> exist
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
> failed to insert into database
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> are you sure you re-installed and the right acc.so is used?
>>
>> The support is for generic PV, if it works with one it should work with
>> any. Which one does work for you?
>>
>> Can you give exact table name as printed in the SQL query? Does it have
>> the parenthesis and the format string?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>> Patch applies fine - it still does not seem to take when using
>> $timef, i..e
>>
>>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>
>>  It looks as if the $timef is not being interpreted and is simply
>> writing to acc'ing i.e. INSERT INTO acc_$timef
>>
>>  Thanks!
>>
>>  Sincerely,
>> Brandon Armstead
>> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I reapplied the patch (cherry-picked from the initial one) to the master
>>> branch.
>>>
>>> Can you test that and see if it works fine?
>>>
>>> You can cherry-picked to your branch, try:
>>>
>>> git pull origin
>>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>>
>>> If it applies fine, compiles and the test go ok, then I will backport as
>>> soon as Juha confirms that was no solid reason in discarding this feature
>>> by his commit.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>
>>> Daniel,
>>>
>>> In my research I saw that commit as well but figured it was some
>>> kind of weird merging error.  Thanks for your time!  Look forward to
>>> hearing back from you guys.
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>>  Hello,
>>>>
>>>> checked the sources and it seems that Juha reverted this feature with
>>>> the commit:
>>>>
>>>>
>>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>>
>>>> I guess it was accidentally, I will ask on devels list.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>>
>>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>
>>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>>> mico...@gmail.com> wrote:
>>>>
>>>>   Hello,
>>>>
>>>> this functionality should be already there. Doesn't work for you?
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>
>>>> Sorry to wake up an old thread.
>>>>
>>>>  However - I am looking to export this $timef function to the param
>>>> initialization for accounting, i.e. db_ta

Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

WORKS!

Thank you.

On Tue, Aug 14, 2012 at 1:22 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> the $timef(...) is evaluated, but you don't have time specifiers there,
> only static letters. You have to use % in front of the letters, as I guess
> from the example:
>
> http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables
>
> Try acc_$timef(%Y%m%d).
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 10:17 PM, Brandon Armstead wrote:
>
> Daniel,
>
>  My apologies - forgot to make clean.  However still no cigar.
>
>  Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
> [km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
> exist
> Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
> failed to insert into database
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> are you sure you re-installed and the right acc.so is used?
>>
>> The support is for generic PV, if it works with one it should work with
>> any. Which one does work for you?
>>
>> Can you give exact table name as printed in the SQL query? Does it have
>> the parenthesis and the format string?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>> Patch applies fine - it still does not seem to take when using
>> $timef, i..e
>>
>>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>>
>>  It looks as if the $timef is not being interpreted and is simply
>> writing to acc'ing i.e. INSERT INTO acc_$timef
>>
>>  Thanks!
>>
>>  Sincerely,
>> Brandon Armstead
>> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> I reapplied the patch (cherry-picked from the initial one) to the master
>>> branch.
>>>
>>> Can you test that and see if it works fine?
>>>
>>> You can cherry-picked to your branch, try:
>>>
>>> git pull origin
>>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>>
>>> If it applies fine, compiles and the test go ok, then I will backport as
>>> soon as Juha confirms that was no solid reason in discarding this feature
>>> by his commit.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>>
>>> Daniel,
>>>
>>> In my research I saw that commit as well but figured it was some
>>> kind of weird merging error.  Thanks for your time!  Look forward to
>>> hearing back from you guys.
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>>  Hello,
>>>>
>>>> checked the sources and it seems that Juha reverted this feature with
>>>> the commit:
>>>>
>>>>
>>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>>
>>>> I guess it was accidentally, I will ask on devels list.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>>
>>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>>
>>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>>> mico...@gmail.com> wrote:
>>>>
>>>>   Hello,
>>>>
>>>> this functionality should be already there. Doesn't work for you?
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>>
>>>> Sorry to wake up an old thread.
>>>>
>>>>  However - I am looking to export this $timef function to the param
>>>> initialization for accounting, i.e. db_table_acc
>>>>
>>>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>>
>>>>  I've looked into completing this mysel

Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Daniel,

My apologies - forgot to make clean.  However still no cigar.

Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Table 'kamailio.acc_Ymd' doesn't
exist
Aug 14 20:16:40 /usr/local/sbin/kamailio[12410]: ERROR: acc [acc.c:405]:
failed to insert into database

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 1:11 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> are you sure you re-installed and the right acc.so is used?
>
> The support is for generic PV, if it works with one it should work with
> any. Which one does work for you?
>
> Can you give exact table name as printed in the SQL query? Does it have
> the parenthesis and the format string?
>
> Cheers,
> Daniel
>
>
> On 8/14/12 9:55 PM, Brandon Armstead wrote:
>
> Daniel,
>
> Patch applies fine - it still does not seem to take when using
> $timef, i..e
>
>  modparam("acc", "db_table_acc", "acc_$timef(Ymd)")
>
>  It looks as if the $timef is not being interpreted and is simply writing
> to acc'ing i.e. INSERT INTO acc_$timef
>
>  Thanks!
>
>  Sincerely,
> Brandon Armstead
> On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> I reapplied the patch (cherry-picked from the initial one) to the master
>> branch.
>>
>> Can you test that and see if it works fine?
>>
>> You can cherry-picked to your branch, try:
>>
>> git pull origin
>> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>>
>> If it applies fine, compiles and the test go ok, then I will backport as
>> soon as Juha confirms that was no solid reason in discarding this feature
>> by his commit.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>>
>> Daniel,
>>
>> In my research I saw that commit as well but figured it was some
>> kind of weird merging error.  Thanks for your time!  Look forward to
>> hearing back from you guys.
>>
>>  Sincerely,
>> Brandon Armstead
>>
>> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>>  Hello,
>>>
>>> checked the sources and it seems that Juha reverted this feature with
>>> the commit:
>>>
>>>
>>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>>
>>> I guess it was accidentally, I will ask on devels list.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>>
>>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>>
>>> Let me also add that I am using 3.2 but I see the commit was quite some
>>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>>
>>> Sent from my iPhone
>>>
>>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>   Hello,
>>>
>>> this functionality should be already there. Doesn't work for you?
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>>
>>> Sorry to wake up an old thread.
>>>
>>>  However - I am looking to export this $timef function to the param
>>> initialization for accounting, i.e. db_table_acc
>>>
>>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>>
>>>  I've looked into completing this myself however I simply am not
>>> familiar enough at this point between the three different modules that it
>>> would take to implement this (acc, dbsr1, pv).
>>>
>>>  Look forward to any help / insight you may be able to provide.
>>>
>>>  Thanks as always!
>>>
>>>  Sincerely,
>>> Brandon Armstead
>>>
>>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>>
>>>>> Hi Daniel,
>>>>>
>>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>>
>>>>>> I don't know what are all the functions you think of, but for the
>>>>>> example provided above, config file does it easy right now. There is a
>>>>>> pseudo-varia

Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Daniel,

   Patch applies fine - it still does not seem to take when using $timef,
i..e

modparam("acc", "db_table_acc", "acc_$timef(Ymd)")

It looks as if the $timef is not being interpreted and is simply writing to
acc'ing i.e. INSERT INTO acc_$timef

Thanks!

Sincerely,
Brandon Armstead
On Tue, Aug 14, 2012 at 12:48 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Hello,
>
> I reapplied the patch (cherry-picked from the initial one) to the master
> branch.
>
> Can you test that and see if it works fine?
>
> You can cherry-picked to your branch, try:
>
> git pull origin
> git cherry-pick -x 95ee0a3ee75556a25f3a9286837a57decf6c3c91
>
> If it applies fine, compiles and the test go ok, then I will backport as
> soon as Juha confirms that was no solid reason in discarding this feature
> by his commit.
>
> Cheers,
> Daniel
>
>
> On 8/14/12 9:21 PM, Brandon Armstead wrote:
>
> Daniel,
>
> In my research I saw that commit as well but figured it was some kind
> of weird merging error.  Thanks for your time!  Look forward to hearing
> back from you guys.
>
>  Sincerely,
> Brandon Armstead
>
> On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> checked the sources and it seems that Juha reverted this feature with the
>> commit:
>>
>>
>> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>>
>> I guess it was accidentally, I will ask on devels list.
>>
>> Cheers,
>> Daniel
>>
>>
>>
>> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>>
>> Let me also add that I am using 3.2 but I see the commit was quite some
>> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>>
>> Sent from my iPhone
>>
>> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla 
>> wrote:
>>
>>   Hello,
>>
>> this functionality should be already there. Doesn't work for you?
>>
>> Cheers,
>> Daniel
>>
>> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>>
>> Sorry to wake up an old thread.
>>
>>  However - I am looking to export this $timef function to the param
>> initialization for accounting, i.e. db_table_acc
>>
>>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>>
>>  I've looked into completing this myself however I simply am not
>> familiar enough at this point between the three different modules that it
>> would take to implement this (acc, dbsr1, pv).
>>
>>  Look forward to any help / insight you may be able to provide.
>>
>>  Thanks as always!
>>
>>  Sincerely,
>> Brandon Armstead
>>
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>>
>>>> Hi Daniel,
>>>>
>>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>>
>>>>> I don't know what are all the functions you think of, but for the
>>>>> example provided above, config file does it easy right now. There is a
>>>>> pseudo-variable that gives broken-time attribute that can be used with
>>>>> avp_check(), iirc, should be:
>>>>>
>>>>> avp_db_load(...);
>>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>>
>>>>> Of course there is the option of doing while loop, but maybe gets to
>>>>> large for desired config file.
>>>>>
>>>> This is pretty much what I had in mind with my new functions/module, but
>>>> I've completely overlooked that PV when searching the docs for this
>>>> feature. Thank you very much for pointing that out!
>>>>
>>>  for sake of public knowledge, just to add on time specific features:
>>> there is also $timef(format) which returns current time attributes based on
>>> strftime specifiers -- its documentation was missing, I just added it.
>>> Also, there is a transformation {s.ftime,format) which can take any integer
>>> variable holding timestamp and return value based on strftime format.
>>>
>>>
>>> Cheers,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>> http://linkedin.com/in/miconda -- http://twitter.co

Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Daniel,

   In my research I saw that commit as well but figured it was some kind of
weird merging error.  Thanks for your time!  Look forward to hearing back
from you guys.

Sincerely,
Brandon Armstead

On Tue, Aug 14, 2012 at 12:04 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Hello,
>
> checked the sources and it seems that Juha reverted this feature with the
> commit:
>
>
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=959ab319903b9625ead7292cc9638a20146e1cca
>
> I guess it was accidentally, I will ask on devels list.
>
> Cheers,
> Daniel
>
>
>
> On 8/14/12 7:34 PM, Brandon Armstead wrote:
>
> Let me also add that I am using 3.2 but I see the commit was quite some
> time ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!
>
> Sent from my iPhone
>
> On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla 
> wrote:
>
>   Hello,
>
> this functionality should be already there. Doesn't work for you?
>
> Cheers,
> Daniel
>
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>
> Sorry to wake up an old thread.
>
>  However - I am looking to export this $timef function to the param
> initialization for accounting, i.e. db_table_acc
>
>  modparam("db_table_acc", "acc_$ftime(Ymd)");
>
>  I've looked into completing this myself however I simply am not familiar
> enough at this point between the three different modules that it would take
> to implement this (acc, dbsr1, pv).
>
>  Look forward to any help / insight you may be able to provide.
>
>  Thanks as always!
>
>  Sincerely,
> Brandon Armstead
>
> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>>
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>>
>>> Hi Daniel,
>>>
>>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>>
>>>> I don't know what are all the functions you think of, but for the
>>>> example provided above, config file does it easy right now. There is a
>>>> pseudo-variable that gives broken-time attribute that can be used with
>>>> avp_check(), iirc, should be:
>>>>
>>>> avp_db_load(...);
>>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>>
>>>> Of course there is the option of doing while loop, but maybe gets to
>>>> large for desired config file.
>>>>
>>> This is pretty much what I had in mind with my new functions/module, but
>>> I've completely overlooked that PV when searching the docs for this
>>> feature. Thank you very much for pointing that out!
>>>
>>  for sake of public knowledge, just to add on time specific features:
>> there is also $timef(format) which returns current time attributes based on
>> strftime specifiers -- its documentation was missing, I just added it.
>> Also, there is a transformation {s.ftime,format) which can take any integer
>> variable holding timestamp and return value based on strftime format.
>>
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - 
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - 
> http://asipto.com/u/kpw
>
>
> --
> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda 
> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - 
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - 
> http://asipto.com/u/kpw
>
>
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Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Let me also add that I am using 3.2 but I see the commit was quite some time 
ago so I have a feeling it wouldn't work in 3.3 either.  Thanks!

Sent from my iPhone

On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla  
wrote:

> Hello,
> 
> this functionality should be already there. Doesn't work for you?
> 
> Cheers,
> Daniel
> 
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>> Sorry to wake up an old thread.
>> 
>> However - I am looking to export this $timef function to the param 
>> initialization for accounting, i.e. db_table_acc
>> 
>> modparam("db_table_acc", "acc_$ftime(Ymd)");
>> 
>> I've looked into completing this myself however I simply am not familiar 
>> enough at this point between the three different modules that it would take 
>> to implement this (acc, dbsr1, pv).
>> 
>> Look forward to any help / insight you may be able to provide.
>> 
>> Thanks as always!
>> 
>> Sincerely,
>> Brandon Armstead
>> 
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla 
>>  wrote:
>> Hello,
>> 
>> 
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>> Hi Daniel,
>> 
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>> I don't know what are all the functions you think of, but for the
>> example provided above, config file does it easy right now. There is a
>> pseudo-variable that gives broken-time attribute that can be used with
>> avp_check(), iirc, should be:
>> 
>> avp_db_load(...);
>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>> 
>> Of course there is the option of doing while loop, but maybe gets to
>> large for desired config file.
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>> for sake of public knowledge, just to add on time specific features: there 
>> is also $timef(format) which returns current time attributes based on 
>> strftime specifiers -- its documentation was missing, I just added it. Also, 
>> there is a transformation {s.ftime,format) which can take any integer 
>> variable holding timestamp and return value based on strftime format.
>> 
>> 
>> Cheers,
>> Daniel
>> 
>> -- 
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>> 
>> 
>> ___
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> -- 
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> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - 
> http://asipto.com/u/katu
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Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Daniel,

   It ONLY works with the $time avp

Sent from my iPhone

On Aug 14, 2012, at 10:27 AM, Daniel-Constantin Mierla  
wrote:

> Hello,
> 
> this functionality should be already there. Doesn't work for you?
> 
> Cheers,
> Daniel
> 
> On 8/14/12 7:16 PM, Brandon Armstead wrote:
>> Sorry to wake up an old thread.
>> 
>> However - I am looking to export this $timef function to the param 
>> initialization for accounting, i.e. db_table_acc
>> 
>> modparam("db_table_acc", "acc_$ftime(Ymd)");
>> 
>> I've looked into completing this myself however I simply am not familiar 
>> enough at this point between the three different modules that it would take 
>> to implement this (acc, dbsr1, pv).
>> 
>> Look forward to any help / insight you may be able to provide.
>> 
>> Thanks as always!
>> 
>> Sincerely,
>> Brandon Armstead
>> 
>> On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla 
>>  wrote:
>> Hello,
>> 
>> 
>> On 12/19/11 7:50 PM, Andreas Granig wrote:
>> Hi Daniel,
>> 
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>> I don't know what are all the functions you think of, but for the
>> example provided above, config file does it easy right now. There is a
>> pseudo-variable that gives broken-time attribute that can be used with
>> avp_check(), iirc, should be:
>> 
>> avp_db_load(...);
>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>> 
>> Of course there is the option of doing while loop, but maybe gets to
>> large for desired config file.
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>> for sake of public knowledge, just to add on time specific features: there 
>> is also $timef(format) which returns current time attributes based on 
>> strftime specifiers -- its documentation was missing, I just added it. Also, 
>> there is a transformation {s.ftime,format) which can take any integer 
>> variable holding timestamp and return value based on strftime format.
>> 
>> 
>> Cheers,
>> Daniel
>> 
>> -- 
>> Daniel-Constantin Mierla -- http://www.asipto.com
>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>> 
>> 
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>> 
> 
> -- 
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> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - 
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - 
> http://asipto.com/u/kpw
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Re: [SR-Users] Suggestions for time utils

[Brandon Armstead]

Sorry to wake up an old thread.

However - I am looking to export this $timef function to the param
initialization for accounting, i.e. db_table_acc

modparam("db_table_acc", "acc_$ftime(Ymd)");

I've looked into completing this myself however I simply am not familiar
enough at this point between the three different modules that it would take
to implement this (acc, dbsr1, pv).

Look forward to any help / insight you may be able to provide.

Thanks as always!

Sincerely,
Brandon Armstead

On Mon, Dec 19, 2011 at 11:55 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

> Hello,
>
>
> On 12/19/11 7:50 PM, Andreas Granig wrote:
>
>> Hi Daniel,
>>
>> On 12/19/2011 07:29 PM, Daniel-Constantin Mierla wrote:
>>
>>> I don't know what are all the functions you think of, but for the
>>> example provided above, config file does it easy right now. There is a
>>> pseudo-variable that gives broken-time attribute that can be used with
>>> avp_check(), iirc, should be:
>>>
>>> avp_db_load(...);
>>> if(avp_check("$time(wday)", "eq/$avp(s:cf_weekday)/g")) { do CF }
>>>
>>> Of course there is the option of doing while loop, but maybe gets to
>>> large for desired config file.
>>>
>> This is pretty much what I had in mind with my new functions/module, but
>> I've completely overlooked that PV when searching the docs for this
>> feature. Thank you very much for pointing that out!
>>
> for sake of public knowledge, just to add on time specific features: there
> is also $timef(format) which returns current time attributes based on
> strftime specifiers -- its documentation was missing, I just added it.
> Also, there is a transformation {s.ftime,format) which can take any integer
> variable holding timestamp and return value based on strftime format.
>
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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Re: [SR-Users] Best tool for crafting sip messages

[Brandon Armstead]

Sipsak

Sent from my iPhone

On Aug 12, 2012, at 10:41 PM, Anton Kvashenkin  wrote:

> Hello, guys. 
> 
> What is the best tool for crafting sip messages for testing purpose? For 
> example, simple REGISTER or INVITE message. I'm new to kamailio (not sip) so 
> it would be great for me to use debbuger module for step-by-step digging into 
> cfg.
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Re: [SR-Users] AVPOPS / TM behavior

[Brandon Armstead]

Klaus,

   I see the following:

AVPs are special variables that are attached to SIP transactions. It is a
list of pairs (name,value). Before the transaction is created, the AVP list
is attached to SIP request. Note that the AVP list works like a stack, last
added value is retrieved first, and there can be many values for same AVP
name, an assignment to the same AVP name does not overwrite old value, it
will add the new value in the list.
While this does *technically* describe the behavior - we may want to
explicitly point out this behavior when spiraling to the same proxy.  I
guess to me its not clear enough based off of this above copied text.
 Unless I am still missing the explanation somewhere else in the text?
 Thanks!

Sincerely,
Brandon Armstead
On Tue, Aug 7, 2012 at 12:00 PM, Klaus Darilion <
klaus.mailingli...@pernau.at> wrote:

> Once you know it you will find it :-)
>
> http://www.kamailio.org/wiki/**cookbooks/3.3.x/**pseudovariables#avps<http://www.kamailio.org/wiki/cookbooks/3.3.x/pseudovariables#avps>
>
> regards
> Klaus
>
>
> On 07.08.2012 18:22, Brandon Armstead wrote:
>
>> Klaus,
>>
>> Thank you for this detailed explanation.  This is essentially what I
>> figured was happening.  I was able to use htable to work around it.
>>
>> I guess however I am still confused as to where there is any public
>> documentation on this specific bit.  Had I've not been working with
>> Kamailio for years I would think this would confuse others.
>>
>> Let me know if it is somewhere else, otherwise I will add it to the
>> Kamailio wiki.
>>
>> Sincerely,
>> Brandon Armstead
>>
>> On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion
>> > <mailto:klaus.mailinglists@**pernau.at>>
>> wrote:
>>
>> AVPs are associated with the transaction. If you "spiral" a request
>> through the same proxy, then for the proxy it is a new transaction.
>> Thus, when processing the request a second time, there is a new
>> transaction and you do not have access to the AVPs of the previous
>> transaction.
>>
>> Workarounds are:
>> - store data in SIP headers and retrieve it later (ugly)
>> - use htable module to store data during transaction 1 and retrieve
>>     it during transaction 2. Therefore you need a known "key" which is
>> identical in this 2 transactions only (e.g. use "$ci$ft" as base for
>> the key).
>>
>> regards
>> Klaus
>>
>>
>>
>>
>>
>> On 07.08.2012 00:27, Brandon Armstead wrote:
>>
>> Hello,
>>
>>  I am curious if there is any documentation on how AVP's
>> processing
>> works in the following scenario below.
>>
>> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once
>> I
>> relay back to the same KAMAILIO proxy (self)?
>>
>> Is there any documentation on why or when this would occur?
>>
>> Is there a better way to handle such a scenario?  i.e. more
>> dynamic
>> internal routing, vs relaying to self.
>>
>> Thanks as always in advance!
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
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>>
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>> >
>>
>>
>>
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Re: [SR-Users] AVPOPS / TM behavior

[Brandon Armstead]

Klaus,

   Thank you for this detailed explanation.  This is essentially what I
figured was happening.  I was able to use htable to work around it.

I guess however I am still confused as to where there is any public
documentation on this specific bit.  Had I've not been working with
Kamailio for years I would think this would confuse others.

Let me know if it is somewhere else, otherwise I will add it to the
Kamailio wiki.

Sincerely,
Brandon Armstead

On Tue, Aug 7, 2012 at 2:32 AM, Klaus Darilion  wrote:

> AVPs are associated with the transaction. If you "spiral" a request
> through the same proxy, then for the proxy it is a new transaction. Thus,
> when processing the request a second time, there is a new transaction and
> you do not have access to the AVPs of the previous transaction.
>
> Workarounds are:
> - store data in SIP headers and retrieve it later (ugly)
> - use htable module to store data during transaction 1 and retrieve it
> during transaction 2. Therefore you need a known "key" which is identical
> in this 2 transactions only (e.g. use "$ci$ft" as base for the key).
>
> regards
> Klaus
>
>
>
>
>
> On 07.08.2012 00:27, Brandon Armstead wrote:
>
>> Hello,
>>
>> I am curious if there is any documentation on how AVP's processing
>> works in the following scenario below.
>>
>> UAC 1 -> KAMAILIO -> KAMAILIO -> DEST
>>
>> It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I
>> relay back to the same KAMAILIO proxy (self)?
>>
>> Is there any documentation on why or when this would occur?
>>
>> Is there a better way to handle such a scenario?  i.e. more dynamic
>> internal routing, vs relaying to self.
>>
>> Thanks as always in advance!
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> __**_
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>>
>>
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[SR-Users] AVPOPS / TM behavior

[Brandon Armstead]

Hello,

   I am curious if there is any documentation on how AVP's processing works
in the following scenario below.

UAC 1 -> KAMAILIO -> KAMAILIO -> DEST

It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I relay
back to the same KAMAILIO proxy (self)?

Is there any documentation on why or when this would occur?

Is there a better way to handle such a scenario?  i.e. more dynamic
internal routing, vs relaying to self.

Thanks as always in advance!

Sincerely,
Brandon Armstead
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Re: [SR-Users] BYE Race Condition

[Brandon Armstead]

Daniel,

   I will try this -- and get back to you.  I noticed the 408 timeout as
well -- and thought that this call flow was strange.  Thanks!

Sincerely,
Brandon Armstead

On Mon, Jul 30, 2012 at 1:03 AM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> first, such race can happen always and it is ok from sip rfc point of
> view. The carrier UA should have received the BYE from the other side and
> close the dialog, then ignore the rest. So it is a broken UA implementation
> imo.
>
> Let's say you just drop the 481, then the BYE will time out (408)? Is the
> carrier UA still complaining? You can make a failure route for BYE and if
> it is 481, then use t_reply("408", "Timeout") if that makes the UA happier.
>
> Cheers,
> Daniel
>
>
> On 7/28/12 8:42 PM, Brandon Armstead wrote:
>
> Hello,
>
> I am running into an issue where there is a race condition happening.
>  I am looking for opinions / ideas on how to handle the following below
> scenario.
>
>  Scenario.
>
>  UAC places an outbound call -> upstream carrier.
>
>  The call is disconnected on both ends at the exact same time,
>
>  UAC -> sends BYE upstream
>
>  Upstream Carrier -> sends BYE downstream
>
>  Upstream 200 OK's the BYE
>
>  UAC sends 481 back to Upstream Carrier for their generated BYE.
>
>  The upstream carrier is complaining about receiving the relayed 481
> responses -- so my first thought was simply to drop() these from relaying
> upstream.
>
>  I am curious how other people are handling this?
>
>  Would you suggest simply dropping the relay from being sent back
> upstream on the 481?
>
>  Would you simply always 200 OK a downstream BYE from trusted carriers
> regardless of UAC response, and create separate transaction to send BYE
> downstream?
>
>  Thank you as always.  Look forward to your thoughts / suggestions /
> ideas.
>
>  Sincerely,
> Brandon Armstead
>
>
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>
> --
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> - http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - 
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - 
> http://asipto.com/u/kpw
>
>
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[SR-Users] BYE Race Condition

[Brandon Armstead]

*** Please forgive if this is a duplicate ***

Hello,

   I am running into an issue where there is a race condition happening.  I
am looking for opinions / ideas on how to handle the following below
scenario.

Scenario.

UAC places an outbound call -> upstream carrier.

The call is disconnected on both ends at the exact same time,

UAC -> sends BYE upstream

Upstream Carrier -> sends BYE downstream

Upstream 200 OK's the BYE

UAC sends 481 back to Upstream Carrier for their generated BYE.

The upstream carrier is complaining about receiving the relayed 481
responses -- so my first thought was simply to drop() these from relaying
upstream.

I am curious how other people are handling this?

Would you suggest simply dropping the relay from being sent back upstream
on the 481?

Would you simply always 200 OK a downstream BYE from trusted carriers
regardless of UAC response, and create separate transaction to send BYE
downstream?

Thank you as always.  Look forward to your thoughts / suggestions / ideas.

Sincerely,
Brandon Armstead
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[SR-Users] BYE Race Condition

[Brandon Armstead]

Hello,

   I am running into an issue where there is a race condition happening.  I
am looking for opinions / ideas on how to handle the following below
scenario.

Scenario.

UAC places an outbound call -> upstream carrier.

The call is disconnected on both ends at the exact same time,

UAC -> sends BYE upstream

Upstream Carrier -> sends BYE downstream

Upstream 200 OK's the BYE

UAC sends 481 back to Upstream Carrier for their generated BYE.

The upstream carrier is complaining about receiving the relayed 481
responses -- so my first thought was simply to drop() these from relaying
upstream.

I am curious how other people are handling this?

Would you suggest simply dropping the relay from being sent back upstream
on the 481?

Would you simply always 200 OK a downstream BYE from trusted carriers
regardless of UAC response, and create separate transaction to send BYE
downstream?

Thank you as always.  Look forward to your thoughts / suggestions / ideas.

Sincerely,
Brandon Armstead
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Re: [SR-Users] How to use regular expression for $var ?

[Brandon Armstead]

Gary,

  Please elaborate on "did not work".  Also have you logged the actual value of 
this var before the if/else statement?

Sent from my iPhone

On Jul 16, 2012, at 7:32 AM, Gary Chen  wrote:

> Kamailio 3.3.0
>  
> I have a variable  $var(s:dst). It can store either a number or IP.
> How do I check to determine whether it is a number of IP?
> I tried the following and it did not work:
> If ($var(s:dst) =~ “^\d+\.\d+\.\d+\.\d+$”){
> It is a IP.
> }else{
> It is a Number
> }
>  
>  
> Thanks
>  
> Gary
>  
>  
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Re: [SR-Users] avp_delete problem [kamailio 1.5]

[Brandon Armstead]

Sven,
  It looks as if it's pushing onto avp stack.  Try doing the assignment as 
follows below.

$avp(test) := "value";

Sent from my iPhone

On Feb 21, 2012, at 11:45 PM, Sven Knoblich  wrote:

> Hello all,
> i am currently confused by using the avpops function avp_delete. When i run 
> avp_delete without the flag-value \g only the last value will be unset (like 
> an undo). Is this the wanted behaviour?
> 
> 
> EXAMPLE:
> 
> $avp(test) = "test"
> if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");}
> 
> $avp(test) = "nooo";
> if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");}
> 
> avp_delete("$avp(test)");
> if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");}
> 
> 
> RESULT:
> 
> result1:test
> result2:nooo
> result3:test
> 
> Could anybody helps me to understand that?
> 
> thanks in advance,
> Sven
> 
> 
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Re: [SR-Users] Double Invite / Issue

[Brandon Armstead]

Daniel,

I believe to have located the issue -- my apologies for not updating
this thread however I wanted to confirm its resolution.  It appears that
there was an email that went out stating to take out km_append_branch /
append_branch from failure route.  It was appending two branches, one when
I updated $ru and another after I did append_branch.  Taking out the
append_branch call resolved the issue.

Sincerely,
Brandon Armstead

On Wed, Dec 14, 2011 at 1:36 AM, Daniel-Constantin Mierla  wrote:

> Hello,
>
>
> On 12/14/11 1:14 AM, Brandon Armstead wrote:
>
>> Hello,
>>
>>I am running into a problem where I am experiencing duplicated
>> INVITE's being originated from Kamailio proxy.
>>
>> U 2011/12/13 23:58:17.543802 KAMAILIO:5060 -> PSTN:5060
>> INVITE sip:URI@PSTN SIP/2.0.
>>
>> U 2011/12/13 23:58:17.543840 KAMAILIO:5060 -> PSTN:5060
>> INVITE sip:URI@PSTN SIP/2.0.
>>
>> As you can see that the timestamp between the two invites is literally a
>> fraction of seconds.
>>
>> There is no failure condition that is causing this with append_branch
>>
>> There is no retransmission timer issue that is causing this (using
>> default timing):
>>
>> modparam("tm", "retr_timer1", 500)
>> modparam("tm", "retr_timer2", 4000)
>>
>> Any thoughts / ideas - as this is causing a race condition in which there
>> is a 200 OK that is being sent back from upstream -- and we are then
>> CANCELING the 2nd INVITE which is essentially causing on overall problem
>> with the call.
>>
>> Thanks for all help / thoughts / input in advance, thanks!
>>
> if there is a CANCEL for second INVITE, then it is parallel forking with
> two branches. If you can provide ngrep with the INVITEs, I can confirm it
> properly. Can you check there are not two contacts in usrloc?
>
> Cheers,
> Daniel
>
> --
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> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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[SR-Users] Double Invite / Issue

[Brandon Armstead]

Hello,

I am running into a problem where I am experiencing duplicated INVITE's
being originated from Kamailio proxy.

U 2011/12/13 23:58:17.543802 KAMAILIO:5060 -> PSTN:5060
INVITE sip:URI@PSTN SIP/2.0.

U 2011/12/13 23:58:17.543840 KAMAILIO:5060 -> PSTN:5060
INVITE sip:URI@PSTN SIP/2.0.

As you can see that the timestamp between the two invites is literally a
fraction of seconds.

There is no failure condition that is causing this with append_branch

There is no retransmission timer issue that is causing this (using default
timing):

modparam("tm", "retr_timer1", 500)
modparam("tm", "retr_timer2", 4000)

Any thoughts / ideas - as this is causing a race condition in which there
is a 200 OK that is being sent back from upstream -- and we are then
CANCELING the 2nd INVITE which is essentially causing on overall problem
with the call.

Thanks for all help / thoughts / input in advance, thanks!

Sincerely,
Brandon Armstead
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Re: [SR-Users] Dialog / BYE

[Brandon Armstead]

Timo,

It is not equivalent.  Essentially what I see happening is the BYE
packet just comes in - however Kamailio does nothing with it.  It is not
relayed - simply nothing at all happens, on a network level I just see the
BYE coming into the Kamailio proxy and zero response or reaction from
Kamailio.

Sincerely,
Brandon Armstead

On Wed, Nov 2, 2011 at 2:04 PM, Timo Reimann  wrote:

> Hey,
>
>
> Am 02.11.2011 um 08:24 schrieb Brandon Armstead:
> >Some interesting results indeed.  If I place the same block of code
> with or without dlg_get first then dlg_bye AFTER loose_route -- kamailio
> ignores the packet.
> >
> > If I place it before loose_route WITH dlg_get FIRST - it works as
> expected.
>
> What do you mean by "kamailio ignores the packet"? Is this equivalent to
> what you described initially regarding dialogs that were terminated
> single-sided only?
>
>
> Cheers,
>
> --Timo
>
>
>
> > On Tue, Nov 1, 2011 at 4:42 PM, Timo Reimann  wrote:
> > Hey Brandon,
> >
> >
> > Am 02.11.2011 um 00:32 schrieb Brandon Armstead:
> > > Thats correct - however I am calling this before loose_route,
> perhaps that is my problem and need for calling dlg_get?  Let me give this
> a go - and I will respond back with my findings, thanks!
> >
> > Yeah, that should be it -- linking to the currently active dialog is
> being implemented as a callback to record-routed messages which takes place
> after a call to loose_route().
> >
> > Looking forward to hearing your findings. :)
> >
> >
> > Cheers,
> >
> > --Timo
> >
> >
> >
> > > On Tue, Nov 1, 2011 at 4:21 PM, Timo Reimann  wrote:
> > > Hey Brandon,
> > >
> > >
> > > Am 01.11.2011 um 23:44 schrieb Brandon Armstead:
> > > > Thank you for the input - I actually just figured out a
> resolution.
> > > >
> > > > Apparently dlg_bye("all") must be called after dlg_get - so what I'm
> doing now is simply checking for request with a special request uri with
> the callid / from tag / to tag in the request that correlates to the call I
> desire to terminate.  I am then feeding $ci, $tt, and $ft into dlg_get and
> then calling dlg_bye("all") - in which Kamailio then proceeds to send BYE
> out to both legs of the call.
> > > >
> > > > I think we can mark this as resolved :).
> > >
> > > OK, very good. :)
> > >
> > > Let me make sure that this isn't a dialog module bug: You are now
> calling dlg_get() followed by dlg_bye() with proper parameters from within
> the Kamailio configuration script? if so, you shouldn't need to call
> dlg_get() in the first place, at least not if you place dlg_bye() after the
> call to loose_route(). Contrary, using certain dialog features (including
> dlg_bye()) from reply routes isn't completely possible yet which could
> explain your findings.
> > >
> > >
> > > Cheers,
> > >
> > > --Timo
> > >
> > >
> > >
> > > > On Tue, Nov 1, 2011 at 3:21 PM, Timo Reimann 
> wrote:
> > > > Hey Brandon,
> > > >
> > > >
> > > > Am 01.11.2011 um 22:48 schrieb Brandon Armstead:
> > > > > I am attempting to tear down a call with a BYE packet
> generated externally (kind of similar to Kamailio fifo dlg_end_dlg).
> > > > >
> > > > > Let me describe what I am trying to do in more depth and then I
> will continue to tell you the problem I think I am experiencing.
> > > > >
> > > > > [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC]
> > > > >
> > > > > So the above layout is the normal call flow / structure of calls
> (incoming when originating from pstn) (outgoing when originating from uac).
> > > > >
> > > > > I then have an "external" host - I am attempting to generate a BYE
> to [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC].
> > > > >
> > > > > So far I am able to get the call to tear down in only a single
> direction (only kill call with PSTN) or (only kill call with UAC).
> > > > >
> > > > > I have not been able to kill both legs of the call.
> > > >
> > > > [snip]
> > > >
> > > >
> > > > > My Question - what am I doing wrong - or what is the best method
> to tackle this task?
> > > >
> > > > If I get you right you are trying to emulate Kamailio's dlg_end_dlg
> functionality without involving Kamailio at all (except for passing forward
> the BYE requests). This makes your question sort of off-topic as processing
> BYE requests is a matter of RFC 3261 alone.
> > > >
> > > > Apart from that, what you could do is take a look at the dialog
> module code and check how it implements dlg_end_dlg. An idea I have what
> you could possibly be doing wrong is not using the CSeq numbers that each
> party expects. I don't have the details at hand but if either CSeq number
> is off your UAs won't accept the BYE request.
> > > >
> > > >
> > > > Cheers,
> > > >
> > > > --Timo
> > >
> >
> >
>
>
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Re: [SR-Users] Dialog / BYE

[Brandon Armstead]

Timo,

   Some interesting results indeed.  If I place the same block of code with
or without dlg_get first then dlg_bye AFTER loose_route -- kamailio ignores
the packet.

If I place it before loose_route WITH dlg_get FIRST - it works as expected.

Sincerely,
Brandon Armstead

On Tue, Nov 1, 2011 at 4:42 PM, Timo Reimann  wrote:

> Hey Brandon,
>
>
> Am 02.11.2011 um 00:32 schrieb Brandon Armstead:
> > Thats correct - however I am calling this before loose_route,
> perhaps that is my problem and need for calling dlg_get?  Let me give this
> a go - and I will respond back with my findings, thanks!
>
> Yeah, that should be it -- linking to the currently active dialog is being
> implemented as a callback to record-routed messages which takes place after
> a call to loose_route().
>
> Looking forward to hearing your findings. :)
>
>
> Cheers,
>
> --Timo
>
>
>
> > On Tue, Nov 1, 2011 at 4:21 PM, Timo Reimann  wrote:
> > Hey Brandon,
> >
> >
> > Am 01.11.2011 um 23:44 schrieb Brandon Armstead:
> > > Thank you for the input - I actually just figured out a resolution.
> > >
> > > Apparently dlg_bye("all") must be called after dlg_get - so what I'm
> doing now is simply checking for request with a special request uri with
> the callid / from tag / to tag in the request that correlates to the call I
> desire to terminate.  I am then feeding $ci, $tt, and $ft into dlg_get and
> then calling dlg_bye("all") - in which Kamailio then proceeds to send BYE
> out to both legs of the call.
> > >
> > > I think we can mark this as resolved :).
> >
> > OK, very good. :)
> >
> > Let me make sure that this isn't a dialog module bug: You are now
> calling dlg_get() followed by dlg_bye() with proper parameters from within
> the Kamailio configuration script? if so, you shouldn't need to call
> dlg_get() in the first place, at least not if you place dlg_bye() after the
> call to loose_route(). Contrary, using certain dialog features (including
> dlg_bye()) from reply routes isn't completely possible yet which could
> explain your findings.
> >
> >
> > Cheers,
> >
> > --Timo
> >
> >
> >
> > > On Tue, Nov 1, 2011 at 3:21 PM, Timo Reimann  wrote:
> > > Hey Brandon,
> > >
> > >
> > > Am 01.11.2011 um 22:48 schrieb Brandon Armstead:
> > > > I am attempting to tear down a call with a BYE packet generated
> externally (kind of similar to Kamailio fifo dlg_end_dlg).
> > > >
> > > > Let me describe what I am trying to do in more depth and then I will
> continue to tell you the problem I think I am experiencing.
> > > >
> > > > [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC]
> > > >
> > > > So the above layout is the normal call flow / structure of calls
> (incoming when originating from pstn) (outgoing when originating from uac).
> > > >
> > > > I then have an "external" host - I am attempting to generate a BYE
> to [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC].
> > > >
> > > > So far I am able to get the call to tear down in only a single
> direction (only kill call with PSTN) or (only kill call with UAC).
> > > >
> > > > I have not been able to kill both legs of the call.
> > >
> > > [snip]
> > >
> > >
> > > > My Question - what am I doing wrong - or what is the best method to
> tackle this task?
> > >
> > > If I get you right you are trying to emulate Kamailio's dlg_end_dlg
> functionality without involving Kamailio at all (except for passing forward
> the BYE requests). This makes your question sort of off-topic as processing
> BYE requests is a matter of RFC 3261 alone.
> > >
> > > Apart from that, what you could do is take a look at the dialog module
> code and check how it implements dlg_end_dlg. An idea I have what you could
> possibly be doing wrong is not using the CSeq numbers that each party
> expects. I don't have the details at hand but if either CSeq number is off
> your UAs won't accept the BYE request.
> > >
> > >
> > > Cheers,
> > >
> > > --Timo
> >
>
>
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Re: [SR-Users] Dialog / BYE

[Brandon Armstead]

Timo,

Thats correct - however I am calling this before loose_route, perhaps
that is my problem and need for calling dlg_get?  Let me give this a go -
and I will respond back with my findings, thanks!

Sincerely,
Brandon Armstead

On Tue, Nov 1, 2011 at 4:21 PM, Timo Reimann  wrote:

> Hey Brandon,
>
>
> Am 01.11.2011 um 23:44 schrieb Brandon Armstead:
> > Thank you for the input - I actually just figured out a resolution.
> >
> > Apparently dlg_bye("all") must be called after dlg_get - so what I'm
> doing now is simply checking for request with a special request uri with
> the callid / from tag / to tag in the request that correlates to the call I
> desire to terminate.  I am then feeding $ci, $tt, and $ft into dlg_get and
> then calling dlg_bye("all") - in which Kamailio then proceeds to send BYE
> out to both legs of the call.
> >
> > I think we can mark this as resolved :).
>
> OK, very good. :)
>
> Let me make sure that this isn't a dialog module bug: You are now calling
> dlg_get() followed by dlg_bye() with proper parameters from within the
> Kamailio configuration script? if so, you shouldn't need to call dlg_get()
> in the first place, at least not if you place dlg_bye() after the call to
> loose_route(). Contrary, using certain dialog features (including
> dlg_bye()) from reply routes isn't completely possible yet which could
> explain your findings.
>
>
> Cheers,
>
> --Timo
>
>
>
> > On Tue, Nov 1, 2011 at 3:21 PM, Timo Reimann  wrote:
> > Hey Brandon,
> >
> >
> > Am 01.11.2011 um 22:48 schrieb Brandon Armstead:
> > > I am attempting to tear down a call with a BYE packet generated
> externally (kind of similar to Kamailio fifo dlg_end_dlg).
> > >
> > > Let me describe what I am trying to do in more depth and then I will
> continue to tell you the problem I think I am experiencing.
> > >
> > > [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC]
> > >
> > > So the above layout is the normal call flow / structure of calls
> (incoming when originating from pstn) (outgoing when originating from uac).
> > >
> > > I then have an "external" host - I am attempting to generate a BYE to
> [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC].
> > >
> > > So far I am able to get the call to tear down in only a single
> direction (only kill call with PSTN) or (only kill call with UAC).
> > >
> > > I have not been able to kill both legs of the call.
> >
> > [snip]
> >
> >
> > > My Question - what am I doing wrong - or what is the best method to
> tackle this task?
> >
> > If I get you right you are trying to emulate Kamailio's dlg_end_dlg
> functionality without involving Kamailio at all (except for passing forward
> the BYE requests). This makes your question sort of off-topic as processing
> BYE requests is a matter of RFC 3261 alone.
> >
> > Apart from that, what you could do is take a look at the dialog module
> code and check how it implements dlg_end_dlg. An idea I have what you could
> possibly be doing wrong is not using the CSeq numbers that each party
> expects. I don't have the details at hand but if either CSeq number is off
> your UAs won't accept the BYE request.
> >
> >
> > Cheers,
> >
> > --Timo
>
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Re: [SR-Users] Dialog / BYE

[Brandon Armstead]

Timo,

Thank you for the input - I actually just figured out a resolution.

Apparently dlg_bye("all") must be called after dlg_get - so what I'm doing
now is simply checking for request with a special request uri with the
callid / from tag / to tag in the request that correlates to the call I
desire to terminate.  I am then feeding $ci, $tt, and $ft into dlg_get and
then calling dlg_bye("all") - in which Kamailio then proceeds to send BYE
out to both legs of the call.

I think we can mark this as resolved :).

Thanks as always!

Sincerely,
Brandon Armstead

On Tue, Nov 1, 2011 at 3:21 PM, Timo Reimann  wrote:

> Hey Brandon,
>
>
> Am 01.11.2011 um 22:48 schrieb Brandon Armstead:
> > I am attempting to tear down a call with a BYE packet generated
> externally (kind of similar to Kamailio fifo dlg_end_dlg).
> >
> > Let me describe what I am trying to do in more depth and then I will
> continue to tell you the problem I think I am experiencing.
> >
> > [PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC]
> >
> > So the above layout is the normal call flow / structure of calls
> (incoming when originating from pstn) (outgoing when originating from uac).
> >
> > I then have an "external" host - I am attempting to generate a BYE to
> [CORE SIP Proxy] and have it go both directions [PSTN] + [UAC].
> >
> > So far I am able to get the call to tear down in only a single direction
> (only kill call with PSTN) or (only kill call with UAC).
> >
> > I have not been able to kill both legs of the call.
>
> [snip]
>
>
> > My Question - what am I doing wrong - or what is the best method to
> tackle this task?
>
> If I get you right you are trying to emulate Kamailio's dlg_end_dlg
> functionality without involving Kamailio at all (except for passing forward
> the BYE requests). This makes your question sort of off-topic as processing
> BYE requests is a matter of RFC 3261 alone.
>
> Apart from that, what you could do is take a look at the dialog module
> code and check how it implements dlg_end_dlg. An idea I have what you could
> possibly be doing wrong is not using the CSeq numbers that each party
> expects. I don't have the details at hand but if either CSeq number is off
> your UAs won't accept the BYE request.
>
>
> Cheers,
>
> --Timo
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[SR-Users] Dialog / BYE

[Brandon Armstead]

Hello,

I am attempting to tear down a call with a BYE packet generated
externally (kind of similar to Kamailio fifo dlg_end_dlg).

Let me describe what I am trying to do in more depth and then I will
continue to tell you the problem I think I am experiencing.

[PSTN SIP Proxy] -> [CORE SIP Proxy] -> [REGISTRAR] -> [UAC]

So the above layout is the normal call flow / structure of calls (incoming
when originating from pstn) (outgoing when originating from uac).

I then have an "external" host - I am attempting to generate a BYE to [CORE
SIP Proxy] and have it go both directions [PSTN] + [UAC].

So far I am able to get the call to tear down in only a single direction
(only kill call with PSTN) or (only kill call with UAC).

I have not been able to kill both legs of the call.

I have tried sending a single BYE and using some dlg_bye("all") magic.

- does not work, only kills one leg of the call.

I have tried sending a BYE message to SIP Proxy for each leg (two BYE).

- does not work, only kills one leg of the call.

If I alternate the later method's BYE packet and only send A or B and not
both Leg A or B will be killed... I think I am running into a race
condition where the dialog is destroyed before the 2nd BYE packet is
processed.

My Question - what am I doing wrong - or what is the best method to tackle
this task?

Thank you for all of your help and time in advance.

Sincerely,
Brandon Armstead
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Re: [SR-Users] Float Comparison

[Brandon Armstead]

Daniel,

   Sorry for taking so long to respond - I am still facing an issue with
this (even in 3.1.4).

I am loading the values from database.

It looks as if the arithmetic operators are not functioning properly when
the data is imported into kamailio from DB - I'm going to do some more
research and give you some specifics.

Sincerely,
Brandon Armstead

On Thu, Jan 20, 2011 at 2:13 AM, Daniel-Constantin Mierla  wrote:

>
>
> On 1/19/11 7:50 AM, Klaus Darilion wrote:
>
>>
>>
>> Am 18.01.2011 21:26, schrieb Brandon Armstead:
>>
>>> Hello,
>>>
>>>Is there anything special that needs to be done for float comparison?
>>>
>>> For example:
>>>
>>> if([5.5 >= 4.3]) 
>>>
>> ^^^ this format is no longer supported starting with 3.0, just skip the
> square brackets, now it is working like in C.
>
>
>
>>> or
>>> if(5.5 > 4.3) 
>>>
>>> The conditional does not seem to be coming back as true like it should?
>>>
>>
>> I have no idea if floating point comparison is supported, but you could
>> multiple the values (e.g. * 1) before comparison
>>
> The pseudo-variables can hold integer or strings. Do you do comparison with
> static values or you load the values in some variables and then compare?
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Jan 24-26, 2011, Irvine, CA, USA
> http://www.asipto.com
>
>
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Re: [SR-Users] Kamailio 3.1.4 Bitwise Operators?

[Brandon Armstead]

Daniel,

   This commit is working perfectly!  Thank you.

Sincerely,
Brandon Armstead

On Tue, Aug 2, 2011 at 3:05 PM, Daniel-Constantin Mierla
wrote:

>  Hello,
>
> some of the bitwise operators were forgotten to be added when we migrated
> to the new core. They were added some time ago, but in master branch since I
> didn't want to commit in stable branch at that moment, because it was just
> few days before 3.1.4, not allowing testing.
>
> I just backported the patch to branch 3.1, so if you fetch the sources from
> GIT now, you should have them and the AND with binary NOT should work
> properly. Let me know if not.
>
> Instead of 'x !~ y' you have to use for now: !(x =~ y).
>
> Cheers,
> Daniel
>
> On 8/2/11 10:44 PM, Brandon Armstead wrote:
>
> Hello,
>
> It seems as if the bitwise operations in Kamailio 3.1.4 are broken?
>  Please correct me if I am doing something wrong - however simple
> evaluations such as:
>
>  $avp(s:my-test-bit) = 10;
>
>  $avp(s:my-test-bit) = (int)$avp(s:my-test-bit) & ~ 2;
>
>  Seems to result in $avp(s:my-test-bit) being equal to "2" rather than "8"
>
>  Any thoughts / Suggestions / Fixes?
>
>  I also wonder if this has any relation to !~ not working?
>
>  Thanks for all of your help in advance!
>
>  P.S. I sent this both to user list and to devel, please forgive me if
> they do not belong in one of the lists.
>
>  Sincerely,
> Brandon Armstead
>
>
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>
>
> --
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> Kamailio Advanced Training, Oct 10-13, Berlin: 
> http://asipto.com/u/kathttp://linkedin.com/in/miconda -- 
> http://twitter.com/miconda
>
>
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[SR-Users] Kamailio 3.1.4 Bitwise Operators?

[Brandon Armstead]

Hello,

   It seems as if the bitwise operations in Kamailio 3.1.4 are broken?
 Please correct me if I am doing something wrong - however simple
evaluations such as:

$avp(s:my-test-bit) = 10;

$avp(s:my-test-bit) = (int)$avp(s:my-test-bit) & ~ 2;

Seems to result in $avp(s:my-test-bit) being equal to "2" rather than "8"

Any thoughts / Suggestions / Fixes?

I also wonder if this has any relation to !~ not working?

Thanks for all of your help in advance!

P.S. I sent this both to user list and to devel, please forgive me if they
do not belong in one of the lists.

Sincerely,
Brandon Armstead
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Re: [SR-Users] Kamailio Multiple Source IP

[Brandon Armstead]

You would use: Force_send_socket method.

Sent from my iPhone

On Jul 26, 2011, at 5:28 PM, Linux Guy  wrote:

> Hi,
>  How can I set a specific source ip for outbound calls when I have multiple 
> Ips that kamailio is listening on.
> 
> For Eg.  I have a table with Caller ID ( Phone Numbers )
> 
> Call Enters Kamailio > Check the table >> If the phone number is found >> 
> SOURCE IP : 1.1.1.1
> 
>   If the 
> phone number is not found >> Source IP : 2.2.2.2
> 
> Kamailio is listening on both 1.1.1.1 and 2.2.2.2  .
> 
> Thanks.
> Linux Guy..
> 
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Re: [SR-Users] ~ operator has no effect

[Brandon Armstead]

Juha,

   Quick correction: $var(test) = $var(test) & ~ 2;

On Tue, May 17, 2011 at 8:13 AM, Juha Heinanen  wrote:

> according to core cookbook:
>
> ~ : bitwise NOT
>
> however, looks like it has no effect.  i have
>
>$var(test) = ~2;
>xlog("L_INFO", "test is <$var(test)>\n");
>
> and i get to syslog:
>
> May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2>
>
> have i misunderstood the operator or is there a bug?
>
> -- juha
>
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Re: [SR-Users] ~ operator has no effect

[Brandon Armstead]

Juha,

Try $var(test) = $var(test) ~ 2;

Sincerely,
Brandon Armstead

On Tue, May 17, 2011 at 8:13 AM, Juha Heinanen  wrote:

> according to core cookbook:
>
> ~ : bitwise NOT
>
> however, looks like it has no effect.  i have
>
>$var(test) = ~2;
>xlog("L_INFO", "test is <$var(test)>\n");
>
> and i get to syslog:
>
> May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2>
>
> have i misunderstood the operator or is there a bug?
>
> -- juha
>
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Re: [SR-Users] Failure Route and Reply Routes

[Brandon Armstead]

Alex,

Thank you for this information!  This is now making some more sense as
to the results I am seeing.  Thanks again!

Sincerely,
Brandon Armstead

On Thu, Feb 17, 2011 at 2:13 PM, Alex Balashov wrote:

> On 02/17/2011 05:07 PM, Brandon Armstead wrote:
>
>> Hello,
>>
>>Is there any specific information / documentation as to what kind
>> of response causes the failure_route to trigger over the reply_route.
>>
>> For example, it seems that 486 response triggers the failure route
>> while as a 408 triggers the reply_route.
>>
>> Perhaps I am over looking something?
>>
>> I would imagine that 5xx and 6xx (final / non provisional responses)
>> would trigger the failure route.
>>
>> While as 4xx would trigger the reply route -- however this does not
>> always seem to be the case in the example of 486 vs 408.
>>
>> Any information / help in advance is greatly appreciated, thanks!
>>
>
> >= 300 responses are considered failures and trigger the failure route.
>  They *also* trigger the reply route first.
>
> >= 200 && < 300 replies trigger the reply route only.
>
> Non-100 1xx provisional replies also trigger the reply route.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
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[SR-Users] Failure Route and Reply Routes

[Brandon Armstead]

Hello,

   Is there any specific information / documentation as to what kind of
response causes the failure_route to trigger over the reply_route.

For example, it seems that 486 response triggers the failure route while as
a 408 triggers the reply_route.

Perhaps I am over looking something?

I would imagine that 5xx and 6xx (final / non provisional responses) would
trigger the failure route.

While as 4xx would trigger the reply route -- however this does not always
seem to be the case in the example of 486 vs 408.

Any information / help in advance is greatly appreciated, thanks!

Sincerely,
Brandon Armstead
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Re: [SR-Users] Can you use a bigint with Kamailio?

[Brandon Armstead]

Lee,

  You will need to cast this to a string / varchar.

Sincerely,
Brandon Armstead

On Sat, Jan 29, 2011 at 5:54 AM, Lee Archer wrote:

>  Hi, I am trying to run dispatcher based on the inbound number called.
> This works fine for some numbers but not all.  Is there a bigint string
> function as this would solve my problems?  The numbers I am receiving are
> 14 digits long which s.int is returning as something completely different
> to what it should be.
>
> Thanks
>
> Lee
>
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[SR-Users] Float Comparison

[Brandon Armstead]

Hello,

   Is there anything special that needs to be done for float comparison?

For example:

if([5.5 >= 4.3]) 

or
if(5.5 > 4.3) 

The conditional does not seem to be coming back as true like it should?

This is simply an example... the precision in the actual case is different
(longer decimal places).


Thank you for all and any help in advance!
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Re: [SR-Users] [sr-dev] set From and To attributes in assignments

[Brandon Armstead]

Very cool, keep up the good work!

On Fri, Aug 13, 2010 at 1:20 AM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> I committed in master branch (to be 3.1.0) code that allows to set uri,
> username, domain and display name for To and From headers using assignments
> to their respective PVs in configuration file. For example:
>
> $fu = "sip:anonym...@invalid";
> $fn = '"Jon Doe"';
> $tU = "+123455678";
>
> Assignment of each such attribute should be done only once, otherwise you
> get concatenated values since it uses the internal lump system. Therefore,
> doing an update is not visible immediately in config unless you do use
> msg_apply_changes().
>
> Also, use it carefully, in case you have sipv1 devices in your network then
> it can break dialog matching. Anyhow changing content of From and To headers
> was possible with remove_hf()/append_hf() or subst() functions, the new
> feature comes just to ease writing the config when one needs it.
>
> Cheers,
> Daniel
>
> --
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> http://www.asipto.com/
>
>
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Re: [SR-Users] Kamailio Transformation Assignment to an AVP

[Brandon Armstead]

Daniel,

When I take your example with the modified regex it works.  I wonder if
there is any association to using / as delimiter versus | and the () match
references?

Sincerely,
Brandon Armstead

On Wed, Jul 28, 2010 at 3:23 AM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> I tested with:
>
> xlog("===+++: [$rb]\n");
> $var(b) = $rb;
> $var(c) = $(rb{re.subst,|.*m=audio [0-9]+ RTP/AVP ([0-9 ]+).+|\1|s});
> # $var(c) = $(var(b){re.subst,|.*m=audio [0-9]+ RTP/AVP ([0-9
> ]+).+|\1|s});
> xlog("===+++: [$rb]\n");
> xlog("==: [$var(c)]\n");
>
> and everything looked fine. Could be something specific for your config or
> environment, what is content-length of your invite?
>
> Cheers,
> Daniel
>
>
>
> On 7/27/10 9:44 PM, bran...@cryy.com wrote:
>
> Daniel,
>
> I've also tried with script vars. I think it has something to do with the
> actual dynamic psuedo variable $rb. Var assignment and mangling does not
> work for me either. I.e. Keep everything the same except $var(test) = $rb
> and so on. Let me know if this makes sense.
>
> Sincerely,
> Brandon Armstead
>
> *Forgot to reply to ALL (resending)
>
> Sent from my Verizon Wireless BlackBerry
> --
> *From: * Daniel-Constantin Mierla  
> *Date: *Tue, 27 Jul 2010 11:50:36 +0200
> *To: *Brandon Armstead 
> *Cc: * ;
>  
> *Subject: *Re: [SR-Users] Kamailio Transformation Assignment to an AVP
>
>  Hello,
>
> interesting, when I troubleshooted before your previous report, I used this
> piece of cfg:
>
> xlog("==\n");
> $var(b) =
> "v=0\r\n"
> "o=- 5 2 IN IP4 192.168.3.100\r\n"
> "s=CounterPath Bria\r\n"
> "c=IN IP4 174.37.XX.XXX\r\n"
> "t=0 0\r\n"
> "m=audio 64192 RTP/AVP 107 0 8 18 101\r\n"
> "a=sendrecv\r\n"
> "a=rtpmap:107 BV32/16000\r\n"
> "a=rtpmap:18 G729/8000\r\n"
> "a=fmtp:18 annexb=yes\r\n"
> "a=rtpmap:101 telephone-event/8000\r\n"
> "a=fmtp:101 0-15\r\n"
> "a=nortpproxy:yes\r\n";
>
> $var(c) = $(var(b){re.subst,|.*m=audio [0-9]+ RTP/AVP ([0-9 ]+).+|\1|s});
> xlog("==: [$var(c)]\n");
>
> And the xlog printed properly the value of $var(c) - meaning that it was
> assigned the right value. I will test with avps next time I have a chance.
>
> Cheers,
> Daniel
>
>
> On 7/26/10 9:11 PM, Brandon Armstead wrote:
>
> Hello,
>
> I have the following transformation on the SDP Body:
>
> $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})
>
> However when I assign this to an AVP, i.e.
>
> $avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9
> ]+)\015\012(.*)$/\3/s});
>
> I receive back a NULL result / transformation "regex does not match" is
> what I receive with high verbose syslog on kamailio.
>
> HOWEVER.
>
> When I do something like this:
>
> xlog("L_INFO", "Payloads Available: $(rb{re.subst,/^(.*)m=audio ([0-9]+)
> RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})");
>
> It executes properly -- transformation regex matches and the payloads are
> displayed as expected.
>
> If anyone can provide any insight as to what I may be doing wrong it would
> be greatly appreciated.
>
> P.S. Kamailio SVN Revision # 2:5906M (1.5.2-notls).
>
> Also I CC'ed devel list (as I do believe this may be a bug).
>
> Thanks!
>
> Sincerely,
> Brandon Armstead
>
>
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>
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>
>
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>
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[SR-Users] Kamailio Transformation Assignment to an AVP

[Brandon Armstead]

Hello,

I have the following transformation on the SDP Body:

$(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})

However when I assign this to an AVP, i.e.

$avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9
]+)\015\012(.*)$/\3/s});

I receive back a NULL result / transformation "regex does not match" is what
I receive with high verbose syslog on kamailio.

HOWEVER.

When I do something like this:

xlog("L_INFO", "Payloads Available: $(rb{re.subst,/^(.*)m=audio ([0-9]+)
RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})");

It executes properly -- transformation regex matches and the payloads are
displayed as expected.

If anyone can provide any insight as to what I may be doing wrong it would
be greatly appreciated.

P.S. Kamailio SVN Revision # 2:5906M (1.5.2-notls).

Also I CC'ed devel list (as I do believe this may be a bug).

Thanks!

Sincerely,
Brandon Armstead
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Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Hello,

To further add more verbose information -- syslog reports back "no
match" for the assignment followed by no right expression and thus its an
error.  While as the xlog "matches" and prints as it should.  (Same exact
transformation code / regex is used lines and no other mangling is
occurring).  Debugging code is Line by Line.  Thanks!

Sincerely,
Brandon Armstead

On Thu, Jul 22, 2010 at 3:08 PM, Brandon Armstead  wrote:

> Daniel,
>
>Too True! haha.
>
> While I've got this topic still open, I am actually having a total freak
> experience going on right now.
>
> I have the following:
>
> xlog("L_INFO", "[$ci] codec / payloads available
> $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9
> ]+)\015\012(.*)$/\3/s})");
>
> Which shows the log correctly and parses out / matches the codecs, i.e (log
> result): [YTFkZjJmNmI3ZWQwZGVlOGQ4MThjNmE0Y2JjODA5ZTU.] codec / payloads
> available 107 0 8 18 101
>
> However... when I assign this to a variable like so:
>
> $avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9
> ]+)\015\012(.*)$/\3/s});
>
> $avp(s:sdp-payloads) is NULL when printing it out?
>
> Any thoughts?
>
> Sincerely,
> Brandon Armstead
>
>
> On Thu, Jul 22, 2010 at 2:53 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>>
>> On 7/22/10 8:59 PM, Brandon Armstead wrote:
>>
>> Hello,
>>
>> Sorry it took me so long to respond back..
>>
>>
>> not a problem, I am one that really knows about unavailability due to
>> traveling or other things. Also, many times it happens that obvious things
>> are "invisible".
>>
>> Cheers,
>> Daniel
>>
>>
>>   I am shocked that I did not find/see that.  That was exactly my issue.
>> :embarrassed:.
>>
>> Thank you!
>>
>> Sincerely,
>> Brandon Armstead
>>
>> On Fri, Jul 16, 2010 at 5:26 AM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> the line you try to match is:
>>>
>>>
>>> m=audio 64192 RTP/AVP 107 0 8 18 101
>>>
>>>  However, the subst does not have rule to match 'RTP/AVP' string:
>>>
>>>
>>> {re.subst,/^(.*)m=audio ([0-9]+) ([0-9 ]+)\015\012(.*)$/\3/s}
>>>
>>>  It is looking for digits and white spaces after m=audio.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 7/5/10 7:38 PM, Brandon Armstead wrote:
>>>
>>> Hello,
>>>
>>> An example $rb body would be:
>>>
>>> v=0#015#012o=- 5 2 IN IP4 192.168.3.100#015#012s=
>>> CounterPath Bria#015#012c=IN IP4 174.37.XX.XXX#015#012t=0
>>> 0#015#012m=audio 64192 RTP/AVP 107 0 8 18
>>> 101#015#012a=sendrecv#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:18
>>> G729/8000#015#012a=fmtp:18 annexb=yes#015#012a=rtpmap:101
>>> telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=nortpproxy:yes#015
>>>
>>> or ngrep version:
>>>
>>> v=0.
>>> o=- 5 2 IN IP4 192.168.3.100.
>>> s=CounterPath Bria.
>>> c=IN IP4 174.37.XX.XXX.
>>> t=0 0.
>>> m=audio 64192 RTP/AVP 107 0 8 18 101.
>>> a=sendrecv.
>>> a=rtpmap:107 BV32/16000.
>>> a=rtpmap:18 G729/8000.
>>> a=fmtp:18 annexb=yes.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=nortpproxy:yes.
>>>
>>> Thanks!
>>>
>>>
>>> On Mon, Jul 5, 2010 at 3:51 AM, Daniel-Constantin Mierla <
>>> mico...@gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>>
>>>> On 7/4/10 9:23 AM, Brandon Armstead wrote:
>>>>
>>>> Hello,
>>>>
>>>> I am trying to match a multi-line psuedo variable, i.e. $rb
>>>>
>>>> However I am wishing to pull out the payload values
>>>>
>>>> i.e.
>>>>
>>>> "0 18 101"
>>>>
>>>> "18 101"
>>>>
>>>> etc.. etc...
>>>>
>>>> I am having trouble matching this.  Any help would be appreciated.
>>>>
>>>> One part that is giving me trouble is that it seems xlog prints out \r\n
>>>> as \015\012
>>>>
>>>>  do you print xlogs to syslog or to terminal?
>>>>
>>>>
>>>>
>>>> I am not able to successfully just pull out the &quo

Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Daniel,

   Too True! haha.

While I've got this topic still open, I am actually having a total freak
experience going on right now.

I have the following:

xlog("L_INFO", "[$ci] codec / payloads available $(rb{re.subst,/^(.*)m=audio
([0-9]+) RTP\/AVP ([0-9 ]+)\015\012(.*)$/\3/s})");

Which shows the log correctly and parses out / matches the codecs, i.e (log
result): [YTFkZjJmNmI3ZWQwZGVlOGQ4MThjNmE0Y2JjODA5ZTU.] codec / payloads
available 107 0 8 18 101

However... when I assign this to a variable like so:

$avp(s:sdp-payloads) = $(rb{re.subst,/^(.*)m=audio ([0-9]+) RTP\/AVP ([0-9
]+)\015\012(.*)$/\3/s});

$avp(s:sdp-payloads) is NULL when printing it out?

Any thoughts?

Sincerely,
Brandon Armstead

On Thu, Jul 22, 2010 at 2:53 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
>
> On 7/22/10 8:59 PM, Brandon Armstead wrote:
>
> Hello,
>
> Sorry it took me so long to respond back..
>
>
> not a problem, I am one that really knows about unavailability due to
> traveling or other things. Also, many times it happens that obvious things
> are "invisible".
>
> Cheers,
> Daniel
>
>
>   I am shocked that I did not find/see that.  That was exactly my issue.
> :embarrassed:.
>
> Thank you!
>
> Sincerely,
> Brandon Armstead
>
> On Fri, Jul 16, 2010 at 5:26 AM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>> Hello,
>>
>> the line you try to match is:
>>
>>
>> m=audio 64192 RTP/AVP 107 0 8 18 101
>>
>>  However, the subst does not have rule to match 'RTP/AVP' string:
>>
>>
>> {re.subst,/^(.*)m=audio ([0-9]+) ([0-9 ]+)\015\012(.*)$/\3/s}
>>
>>  It is looking for digits and white spaces after m=audio.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 7/5/10 7:38 PM, Brandon Armstead wrote:
>>
>> Hello,
>>
>> An example $rb body would be:
>>
>> v=0#015#012o=- 5 2 IN IP4 192.168.3.100#015#012s=
>> CounterPath Bria#015#012c=IN IP4 174.37.XX.XXX#015#012t=0 0#015#012m=audio
>> 64192 RTP/AVP 107 0 8 18 101#015#012a=sendrecv#015#012a=rtpmap:107
>> BV32/16000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
>> annexb=yes#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
>> 0-15#015#012a=nortpproxy:yes#015
>>
>> or ngrep version:
>>
>> v=0.
>> o=- 5 2 IN IP4 192.168.3.100.
>> s=CounterPath Bria.
>> c=IN IP4 174.37.XX.XXX.
>> t=0 0.
>> m=audio 64192 RTP/AVP 107 0 8 18 101.
>> a=sendrecv.
>> a=rtpmap:107 BV32/16000.
>> a=rtpmap:18 G729/8000.
>> a=fmtp:18 annexb=yes.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=nortpproxy:yes.
>>
>> Thanks!
>>
>>
>> On Mon, Jul 5, 2010 at 3:51 AM, Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>>
>>> On 7/4/10 9:23 AM, Brandon Armstead wrote:
>>>
>>> Hello,
>>>
>>> I am trying to match a multi-line psuedo variable, i.e. $rb
>>>
>>> However I am wishing to pull out the payload values
>>>
>>> i.e.
>>>
>>> "0 18 101"
>>>
>>> "18 101"
>>>
>>> etc.. etc...
>>>
>>> I am having trouble matching this.  Any help would be appreciated.
>>>
>>> One part that is giving me trouble is that it seems xlog prints out \r\n
>>> as \015\012
>>>
>>>  do you print xlogs to syslog or to terminal?
>>>
>>>
>>>
>>> I am not able to successfully just pull out the "payloads"
>>>
>>> I have tried many different variations, however here is one of my latest:
>>>
>>> xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9
>>> ]+)\015\012(.*)$/\3/s})");
>>>
>>>
>>>  Can you paste the body you worked on and the output you got? Will help
>>> understanding what happens and maybe give some hits, being easy to reproduce
>>> and test ourselves.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> Any help / input is greatly appreciated, thank you ahead of time!
>>>
>>> Happy 4th of July (for those who celebrate)
>>>
>>> Sincerely,
>>> Brandon Armstead
>>>
>>>
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>>>
>>> --
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>>>
>>>
>>
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>>
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>>
>>
>
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>
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Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Hello,

Sorry it took me so long to respond back..  I am shocked that I did not
find/see that.  That was exactly my issue.  :embarrassed:.

Thank you!

Sincerely,
Brandon Armstead

On Fri, Jul 16, 2010 at 5:26 AM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> the line you try to match is:
>
>
> m=audio 64192 RTP/AVP 107 0 8 18 101
>
> However, the subst does not have rule to match 'RTP/AVP' string:
>
>
> {re.subst,/^(.*)m=audio ([0-9]+) ([0-9 ]+)\015\012(.*)$/\3/s}
>
> It is looking for digits and white spaces after m=audio.
>
> Cheers,
> Daniel
>
>
> On 7/5/10 7:38 PM, Brandon Armstead wrote:
>
> Hello,
>
> An example $rb body would be:
>
> v=0#015#012o=- 5 2 IN IP4 192.168.3.100#015#012s=
> CounterPath Bria#015#012c=IN IP4 174.37.XX.XXX#015#012t=0 0#015#012m=audio
> 64192 RTP/AVP 107 0 8 18 101#015#012a=sendrecv#015#012a=rtpmap:107
> BV32/16000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
> annexb=yes#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
> 0-15#015#012a=nortpproxy:yes#015
>
> or ngrep version:
>
> v=0.
> o=- 5 2 IN IP4 192.168.3.100.
> s=CounterPath Bria.
> c=IN IP4 174.37.XX.XXX.
> t=0 0.
> m=audio 64192 RTP/AVP 107 0 8 18 101.
> a=sendrecv.
> a=rtpmap:107 BV32/16000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=yes.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=nortpproxy:yes.
>
> Thanks!
>
>
> On Mon, Jul 5, 2010 at 3:51 AM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>>
>> On 7/4/10 9:23 AM, Brandon Armstead wrote:
>>
>> Hello,
>>
>> I am trying to match a multi-line psuedo variable, i.e. $rb
>>
>> However I am wishing to pull out the payload values
>>
>> i.e.
>>
>> "0 18 101"
>>
>> "18 101"
>>
>> etc.. etc...
>>
>> I am having trouble matching this.  Any help would be appreciated.
>>
>> One part that is giving me trouble is that it seems xlog prints out \r\n
>> as \015\012
>>
>>  do you print xlogs to syslog or to terminal?
>>
>>
>>
>> I am not able to successfully just pull out the "payloads"
>>
>> I have tried many different variations, however here is one of my latest:
>>
>> xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9
>> ]+)\015\012(.*)$/\3/s})");
>>
>>
>>  Can you paste the body you worked on and the output you got? Will help
>> understanding what happens and maybe give some hits, being easy to reproduce
>> and test ourselves.
>>
>> Cheers,
>> Daniel
>>
>>
>> Any help / input is greatly appreciated, thank you ahead of time!
>>
>> Happy 4th of July (for those who celebrate)
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://www.asipto.com/
>>
>>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com/
>
>
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Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Hey guys,

   I am still hitting my head against the wall with this one.

It seems that re.substr does not interpret any of the following:

\r, \n, \012, \015, etc.

There is no way for me to match control characters.

If you guys know what I am doing wrong it would be very much appreciated
with any input you can provide, thanks!

Sincerely,
Brandon Armstead
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Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Hello,

Please do let me know if I am leaving any thing out that will make
troubleshooting this issue easier.  I am really hitting a wall with an
efficient method to tackle this problem.  All help is appreciated, thanks as
always guys!

Sincerely,
Brandon Armstead

On Mon, Jul 5, 2010 at 12:38 PM, Brandon Armstead  wrote:

> Hello,
>
> An example $rb body would be:
>
> v=0#015#012o=- 5 2 IN IP4 192.168.3.100#015#012s=
> CounterPath Bria#015#012c=IN IP4 174.37.XX.XXX#015#012t=0 0#015#012m=audio
> 64192 RTP/AVP 107 0 8 18 101#015#012a=sendrecv#015#012a=rtpmap:107
> BV32/16000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
> annexb=yes#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
> 0-15#015#012a=nortpproxy:yes#015
>
> or ngrep version:
>
> v=0.
> o=- 5 2 IN IP4 192.168.3.100.
> s=CounterPath Bria.
> c=IN IP4 174.37.XX.XXX.
> t=0 0.
> m=audio 64192 RTP/AVP 107 0 8 18 101.
> a=sendrecv.
> a=rtpmap:107 BV32/16000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=yes.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=nortpproxy:yes.
>
> Thanks!
>
>
> On Mon, Jul 5, 2010 at 3:51 AM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>>
>> On 7/4/10 9:23 AM, Brandon Armstead wrote:
>>
>> Hello,
>>
>> I am trying to match a multi-line psuedo variable, i.e. $rb
>>
>> However I am wishing to pull out the payload values
>>
>> i.e.
>>
>> "0 18 101"
>>
>> "18 101"
>>
>> etc.. etc...
>>
>> I am having trouble matching this.  Any help would be appreciated.
>>
>> One part that is giving me trouble is that it seems xlog prints out \r\n
>> as \015\012
>>
>> do you print xlogs to syslog or to terminal?
>>
>>
>>
>> I am not able to successfully just pull out the "payloads"
>>
>> I have tried many different variations, however here is one of my latest:
>>
>> xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9
>> ]+)\015\012(.*)$/\3/s})");
>>
>>
>> Can you paste the body you worked on and the output you got? Will help
>> understanding what happens and maybe give some hits, being easy to reproduce
>> and test ourselves.
>>
>> Cheers,
>> Daniel
>>
>>
>> Any help / input is greatly appreciated, thank you ahead of time!
>>
>> Happy 4th of July (for those who celebrate)
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierlahttp://www.asipto.com/
>>
>>
>
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Re: [SR-Users] Regex Transformations

[Brandon Armstead]

Hello,

An example $rb body would be:

v=0#015#012o=- 5 2 IN IP4 192.168.3.100#015#012s=
CounterPath Bria#015#012c=IN IP4 174.37.XX.XXX#015#012t=0 0#015#012m=audio
64192 RTP/AVP 107 0 8 18 101#015#012a=sendrecv#015#012a=rtpmap:107
BV32/16000#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18
annexb=yes#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=nortpproxy:yes#015

or ngrep version:

v=0.
o=- 5 2 IN IP4 192.168.3.100.
s=CounterPath Bria.
c=IN IP4 174.37.XX.XXX.
t=0 0.
m=audio 64192 RTP/AVP 107 0 8 18 101.
a=sendrecv.
a=rtpmap:107 BV32/16000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=nortpproxy:yes.

Thanks!


On Mon, Jul 5, 2010 at 3:51 AM, Daniel-Constantin Mierla
wrote:

>  Hello,
>
>
> On 7/4/10 9:23 AM, Brandon Armstead wrote:
>
> Hello,
>
> I am trying to match a multi-line psuedo variable, i.e. $rb
>
> However I am wishing to pull out the payload values
>
> i.e.
>
> "0 18 101"
>
> "18 101"
>
> etc.. etc...
>
> I am having trouble matching this.  Any help would be appreciated.
>
> One part that is giving me trouble is that it seems xlog prints out \r\n as
> \015\012
>
> do you print xlogs to syslog or to terminal?
>
>
>
> I am not able to successfully just pull out the "payloads"
>
> I have tried many different variations, however here is one of my latest:
>
> xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9
> ]+)\015\012(.*)$/\3/s})");
>
>
> Can you paste the body you worked on and the output you got? Will help
> understanding what happens and maybe give some hits, being easy to reproduce
> and test ourselves.
>
> Cheers,
> Daniel
>
>
> Any help / input is greatly appreciated, thank you ahead of time!
>
> Happy 4th of July (for those who celebrate)
>
> Sincerely,
> Brandon Armstead
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierlahttp://www.asipto.com/
>
>
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[SR-Users] Regex Transformations

[Brandon Armstead]

Hello,

I am trying to match a multi-line psuedo variable, i.e. $rb

However I am wishing to pull out the payload values

i.e.

"0 18 101"

"18 101"

etc.. etc...

I am having trouble matching this.  Any help would be appreciated.

One part that is giving me trouble is that it seems xlog prints out \r\n as
\015\012

I am not able to successfully just pull out the "payloads"

I have tried many different variations, however here is one of my latest:

xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+) ([0-9
]+)\015\012(.*)$/\3/s})");

Any help / input is greatly appreciated, thank you ahead of time!

Happy 4th of July (for those who celebrate)

Sincerely,
Brandon Armstead
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Re: [SR-Users] [Kamailio-Users] PCRE Regex / Transformations Bug?

[Brandon Armstead]

Daniel,

Here is the XLOG output, the top log is the unmodified version and the
bottom is the modified version.

xlog("L_INFO", "[$ci] $rb") OUTPUTS:

Apr 29 20:17:23 sip-core02 /sbin/kamailio[23550]: [
6db72a2f-7e263...@192.168.1.75] v=0#015#012o=- 24986155 24986155 IN IP4
99.21.137.236#015#012s=-#015#012c=IN IP4 174.37.45.134#015#012t=0
0#015#012m=audio 55630 RTP/AVP 0 2 4 8 18 96 97 98 101#015#012a=rtpmap:0
PCMU/8000#015#012a=rtpmap:2 G726-32/8000#015#012a=rtpmap:4
G723/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:18
G729a/8000#015#012a=rtpmap:96 G726-40/8000#015#012a=rtpmap:97
G726-24/8000#015#012a=rtpmap:98 G726-16/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=ptime:30#015#012a=sendrecv#015#012a=nortpproxy:yes#015

xlog("L_INFO", "[$ci] $(rb{re.subst,/^(.*)m=audio ([0-9]+)(.*)$/\2/})")
OUTPUTS:

Apr 29 20:17:23 sip-core02 /sbin/kamailio[23550]: [
6db72a2f-7e263...@192.168.1.75] v=0#015#012o=- 24986155 24986155 IN IP4
99.21.137.236#015#012s=-#015#012c=IN IP4 174.37.45.134#015#012t=0
0#015#01255630#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:2
G726-32/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:18 G729a/8000#015#012a=rtpmap:96
G726-40/8000#015#012a=rtpmap:97 G726-24/8000#015#012a=rtpmap:98
G726-16/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=ptime:30#015#012a=sendrecv#015#012a=nortpproxy:yes#015

As for the actual INVITE / SDP BODY:

v=0.
o=- 24986155 24986155 IN IP4 99.21.XXX.XXX.
s=-.
c=IN IP4 174.37.XX.XXX.
t=0 0.
m=audio 55630 RTP/AVP 0 2 4 8 18 96 97 98 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729a/8000.
a=rtpmap:96 G726-40/8000.
a=rtpmap:97 G726-24/8000.
a=rtpmap:98 G726-16/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.
a=nortpproxy:yes.

Let me know if this is sufficient or if there is anything else I can
provide.  Thanks!

Sincerely,
Brandon Armstead

On Wed, Apr 28, 2010 at 3:54 PM, Daniel-Constantin Mierla  wrote:

>  Hello,
>
> can you paste here the sip message and the result of the substitution? It
> will help to troubleshoot if is something wrong there.
>
> Cheers,
> Daniel
>
>
>
> On 4/27/10 2:35 AM, Brandon Armstead wrote:
>
> Hello All,
>
>   Correction, it seems both the last supplied regex and xlog("L_INFO",
> "[$ci] m=audio $(rb{re.subst,/(.*)m=audio ([0-9]+) (.*)/\2/})"); return the
> same invalid results.
>
> Thanks again!
>
> Sincerely,
> Brandon Armstead
>
> On Mon, Apr 26, 2010 at 7:33 PM, Brandon Armstead wrote:
>
>> Hello All,
>>
>>I hate to dig-up this older mailing list entry.  However I am some
>> additional trouble with what I believe is a completely posix-only regex.
>>
>> xlog("L_INFO", "[$ci] m=audio $(rb{re.subst,/(.*)m=audio(.*)/\2/})");
>>
>> I would expect to give me the port from the m=audio line, however it does
>> not -- it simply removes m=audio.
>>
>> Any thoughts / ideas / suggestions?
>>
>> Sincerely,
>> Brandon Armstead
>>
>>
>> On Wed, Dec 23, 2009 at 9:31 PM, Iñaki Baz Castillo wrote:
>>
>>> El Miércoles, 23 de Diciembre de 2009, Daniel-Constantin Mierla escribió:
>>> > the re.subst transformation uses Posix regexp for matching, only the
>>> > format of the command is perl-like. I implemented this because textops
>>> > has no dependency of extra libraries than core and I wanted to keep it
>>> so.
>>> >
>>> > Maybe is good to add a pcre.subst transformation in regex module to be
>>> > able to use extended regexp formats given by libpcre.
>>>
>>>  I¡ll try to implement it when I get some spare time :)
>>>
>>>
>>> --
>>> Iñaki Baz Castillo 
>>>
>>> ___
>>>  Kamailio (OpenSER) - Users mailing list
>>> us...@lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>
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>
>
> --
> Daniel-Constantin Mierla
> * http://www.asipto.com/
>
> * http://twitter.com/miconda
> * http://www.linkedin.com/in/danielconstantinmierla
>
>
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Re: [SR-Users] [Kamailio-Users] PCRE Regex / Transformations Bug?

[Brandon Armstead]

Hello All,

  Correction, it seems both the last supplied regex and xlog("L_INFO",
"[$ci] m=audio $(rb{re.subst,/(.*)m=audio ([0-9]+) (.*)/\2/})"); return the
same invalid results.

Thanks again!

Sincerely,
Brandon Armstead

On Mon, Apr 26, 2010 at 7:33 PM, Brandon Armstead  wrote:

> Hello All,
>
>I hate to dig-up this older mailing list entry.  However I am some
> additional trouble with what I believe is a completely posix-only regex.
>
> xlog("L_INFO", "[$ci] m=audio $(rb{re.subst,/(.*)m=audio(.*)/\2/})");
>
> I would expect to give me the port from the m=audio line, however it does
> not -- it simply removes m=audio.
>
> Any thoughts / ideas / suggestions?
>
> Sincerely,
> Brandon Armstead
>
>
> On Wed, Dec 23, 2009 at 9:31 PM, Iñaki Baz Castillo  wrote:
>
>> El Miércoles, 23 de Diciembre de 2009, Daniel-Constantin Mierla escribió:
>> > the re.subst transformation uses Posix regexp for matching, only the
>> > format of the command is perl-like. I implemented this because textops
>> > has no dependency of extra libraries than core and I wanted to keep it
>> so.
>> >
>> > Maybe is good to add a pcre.subst transformation in regex module to be
>> > able to use extended regexp formats given by libpcre.
>>
>> I¡ll try to implement it when I get some spare time :)
>>
>>
>> --
>> Iñaki Baz Castillo 
>>
>> ___
>> Kamailio (OpenSER) - Users mailing list
>> us...@lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>
>
>
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Re: [SR-Users] [Kamailio-Users] PCRE Regex / Transformations Bug?

[Brandon Armstead]

Hello All,

   I hate to dig-up this older mailing list entry.  However I am some
additional trouble with what I believe is a completely posix-only regex.

xlog("L_INFO", "[$ci] m=audio $(rb{re.subst,/(.*)m=audio(.*)/\2/})");

I would expect to give me the port from the m=audio line, however it does
not -- it simply removes m=audio.

Any thoughts / ideas / suggestions?

Sincerely,
Brandon Armstead

On Wed, Dec 23, 2009 at 9:31 PM, Iñaki Baz Castillo  wrote:

> El Miércoles, 23 de Diciembre de 2009, Daniel-Constantin Mierla escribió:
> > the re.subst transformation uses Posix regexp for matching, only the
> > format of the command is perl-like. I implemented this because textops
> > has no dependency of extra libraries than core and I wanted to keep it
> so.
> >
> > Maybe is good to add a pcre.subst transformation in regex module to be
> > able to use extended regexp formats given by libpcre.
>
> I¡ll try to implement it when I get some spare time :)
>
>
> --
> Iñaki Baz Castillo 
>
> ___
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> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
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