Re: [SR-Users] Problem using ds_is_from_list in failure_route
Hi Joel,
On 21/09/16 00:37, Joel Serrano | VOZELIA wrote:
> I managed to solve this problem using ds_is_from_list in failure_route like
> this:
>
> ds_is_from_list("8201", "3", "sip:$T_rpl($si):$T_rpl($sp)"))
>
> But I don't understand why I needed to do it that way.
>
> I would still like anyone to explain if they know the reason!
per my understanding in the failure route you are in the context of the
original INVITE message, so without explicitly telling ds_is_from_list
the IP address it is checking based on the source IP of INVITE.
Andrew
___
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
Re: [SR-Users] should the AVPs be available in CANCEL?
Alex Balashov wrote: > Strictly speaking, CANCEL is a different request, and accordingly, a > different transaction. > > However, you should be able to access INVITE transaction data from the > failure_route triggered in connection with transaction cancellation. Thanks, Alex! Andrew ___ 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
[SR-Users] should the AVPs be available in CANCEL?
Hi all, it seems that the AVPs not available when when processing CANCEL message, even though they have been set for this transaction initially. Is this the expected behavior? P.S. kamailio 4.3.4 Andrew ___ 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
[SR-Users] uac "restore_mode" is auto: effective when using core variables?
Per my understanding the uac module stores the "vsf" parameter in
Record-Route and should be able to update the From/To URIs automatically
in all in-dialog requests that carry this parameter.
http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode
“auto” - all sequential requests and replies will be automatically
updated based on stored original URI. For this option you have to set
“modparam("rr", "append_fromtag", 1)”.
What makes me wonder is: does that only work in From/To was changed by
uac_replace_from()/uac_replace_to() or also when assigning directly to
the $fu and $tu variables? I'm changing those variables and I am using
restore_mode auto but that does not change anything in in-dialog ACK.
I presume this is expected behavior because I'm not using uac when
assigning to variables, isn't it?
Thanks.
Andrew
___
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
Re: [SR-Users] Append to reply in default action of failure_route
Thanks Dmitri, I have to look closer at it, but for me it breaks
parallel forking, e.g. I need to reply only when it's the last branch.
Andrew
On 12/22/2015 05:46 PM, Dmitri Savolainen wrote:
> Andrew, I use smth like this for adding header to any response
>
> request_route{
>
> if (is_method("INVITE")){
> t_check_trans();
> t_on_failure("INV_FAIL");
> t_relay("mysip", "5060");
> }
>
> }
>
>
> failure_route[INV_FAIL] {
> xlog("L_INFO", "failure_route code: $T_reply_code; reason:
> $T_reply_reason;");
> append_to_reply("MyField: my_field_content\r\n");
> send_reply("$T_reply_code","$T_reply_reason");
> exit;
>
> }
___
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
Re: [SR-Users] why msg_apply_changes() from REQUEST_ROUTE only
Hi, I have a similar issue with kamailio 4.3.4. I want to append a header to external 486 or 603 reply. If I got it right I should call append_hf, not append_to_reply which is for locally generated replies. I've added append_hf() and in the end of failure route call exit but for some reason that header does not appear on the wire. So I have added msg_apply_changes() in failure route to see if it changes something for me. 1st surprise is that kamailio actually started with this config. 2nd surprise is that is still doesn't work and throws an error which I didn't see before: Dec 22 16:01:34 sp1 (local7.info) proxy[23957]: INFO:
Re: [SR-Users] Append to reply in default action of failure_route
Hello Efelin,
I stumbled upon the issue you described here. Have you been able to find
a solution? I've tried to play with t_reply(), but no luck so far.
Regards,
Andrew
On 11/18/2013 10:00 AM, Efelin Novak wrote:
> Hi,
>
> I would like to append a header to a 'winning' negative reply in
> failure_route and let the Kamailio do the default action (state fully
> forward the winning reply).
>
> When I use append_to_reply("Foo: bar\r\n"); and then call exit; in
> failure_route nothing is appended.
> When I use same append_to_reply then t_relay("505","Error"); and exit;
> the header is appended.
> When I use append and t_reply with dialog modul turned on I got a bug
> I'm solving here '[SR-Users] t_reply in failure route with dialog module'.
>
> So my question is how to put a header into a reply when I don't want to
> alter its code or text?
>
> I'm using Kamilio 4.0.4 on Debian 7.1
>
> Thanks for an answer
>
> Efelin
___
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
[SR-Users] registered4() is not adding xavp with details of the record
Hi,
I was trying to use registered("location", "$ru", 0, 1)
Last parameter is the flag according to
http://kamailio.org/docs/modules/stable/modules/registrar.html#registrar.f.registered
flag values is as follows:
1 - set xavp_rcd with value from matched contact
But I'm getting NULL instead of ruid.. While the same works after
lookup("location").
So I took a quick look into the code and that confirms that registered4
does not add the xavp with details of the record (ruid), i.e. it does
not do what the lookup_helper does.
Is this done on purpose or an oversight? While fixing this it might be
reasonable to introduce a new function for setting the XAVPs and call it
from lookup and registered4 functions, especially since we are going to
extend the attributes list beyond just ruid, but right now I'm
struggling just to understanding how the XAVP should be built.. Ideas?
Thanks,
Andrew
___
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
Re: [SR-Users] usage of t_flush_flags() in event_route
Daniel,
I'm reviving this old thread..
On 05/27/2015 01:40 PM, Andrew Pogrebennyk wrote:
thanks for the answer, that's what I was thinking - maybe the flags do
not persist or are destroyed after the per-branch failure route.
However, the t_flush_flags description says this function can be used in
any route, so in should be fixed in the long term.. Let me check if I
can come up with some workaround in config script for the time being.
I didn't manage to come up with any workaround so far.
The problem appears if the initial call leg coming from, let's say, the
application server didn't have the accounting flags set. Long story put
short, if the UA responds with the 302 response and we are going to
process that, we want to create an acc record for the new target from
302 message because this call may incur additional costs.
I'm setting the accounting flags and even calling t_flush_flags() but
that doesn't work (no accounting record for INVITE with Call-ID:
rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1).
if($var(redirected_forward) == 1)
{
setflag(FLAG_ACC_FAILED);
setflag(FLAG_ACC_DB);
t_flush_flags();
}
From attached log:
root@sp2:~# grep -i acc kamailio.log
Jun 17 11:49:38 sp2 proxy[12702]: INFO: script: Set callee dialogs: user,
account to 'e59611e6-01de-424c-a04f-e977409c54f6/64' -
R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: INFO: script: Set callee dialogs:
totaluser, totalaccount to 'e59611e6-01de-424c-a04f-e977409c54f6/64' -
R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: INFO: script: Set caller dialogs:
totaluser, totaluserout, totalaccount, totalaccountout to
'0214786f-eb1c-4865-95ba-65d8c1c0bf32/48' - R=sip:te...@demo.mylocal.com
ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: INFO: script: Set caller dialogs: user,
userout, account, accountout to '0214786f-eb1c-4865-95ba-65d8c1c0bf32/48' -
R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: NOTICE: script: Setting acc source-leg
for uuid '0214786f-eb1c-4865-95ba-65d8c1c0bf32':
'0214786f-eb1c-4865-95ba-65d8c1c0bf32|phone2|xxx.demo.mylocal.com|43221000202|||48|||0|call|77.244.249.126|1434534578.107169|||'
- R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: NOTICE: script: ++ 2. ACC flag is set
- R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12702]: NOTICE: script: Setting acc
destination-leg for uuid 'e59611e6-01de-424c-a04f-e977409c54f6':
'0||comx|64|000439911|e59611e6-01de-424c-a04f-e977409c54f6|test1|demo.mylocal.com|439911|xxx.demo.mylocal.com|0|||'
- R=sip:te...@demo.mylocal.com ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV
Jun 17 11:49:38 sp2 proxy[12695]: DEBUG: acc [acc_logic.c:615]: tmcb_func():
acc callback called for t(0x7f421c8da308) event type 2, reply code 100
Jun 17 11:49:38 sp2 proxy[12698]: INFO: script: Skip accounting for call
from PBX to device - R=sip:te...@demo.mylocal.com
ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
^^^
Jun 17 11:49:38 sp2 proxy[12699]: DEBUG: acc [acc_logic.c:615]: tmcb_func():
acc callback called for t(0x7f421c8da308) event type 2, reply code 180
Jun 17 11:49:38 sp2 proxy[12699]: DEBUG: acc [acc_logic.c:615]: tmcb_func():
acc callback called for t(0x7f421c8da308) event type 512, reply code 180
Jun 17 11:49:40 sp2 proxy[12695]: NOTICE: script: -- 1. ACC flag is NOT
set - R=sip:test1@10.15.20.112:5060 ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
Jun 17 11:49:40 sp2 proxy[12695]: INFO: script: Set callee dialogs: user,
account to '72d04db7-6fd2-47c3-95a7-5685a5297715/64' -
R=sip:te...@demo.mylocal.com;alias=77.244.249.126~6774~1
ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
Jun 17 11:49:40 sp2 proxy[12695]: INFO: script: Set callee dialogs:
totaluser, totalaccount to '72d04db7-6fd2-47c3-95a7-5685a5297715/64' -
R=sip:te...@demo.mylocal.com;alias=77.244.249.126~6774~1
ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
Jun 17 11:49:40 sp2 proxy[12695]: INFO: script: Set caller dialogs:
totaluser, totaluserout, totalaccount, totalaccountout to
'e59611e6-01de-424c-a04f-e977409c54f6/64' - R=sip:25704386@10.10.8.52:46637
ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
Jun 17 11:49:40 sp2 proxy[12695]: INFO: script: Set caller dialogs: user,
userout, account, accountout to 'e59611e6-01de-424c-a04f-e977409c54f6/64' -
R=sip:25704386@10.10.8.52:46637 ID=rOemTINsCbQspe1Vlje9nAK15SNy0DdV_pbx-1
Jun 17 11:49:40 sp2 proxy[12695]: NOTICE: script: Setting acc source-leg
for uuid 'e59611e6-01de-424c-a04f-e977409c54f6':
'e59611e6-01de-424c-a04f-e977409c54f6|test1|xxx.demo.mylocal.com|phone2||comx|64|||null|cfb|77.244.249.126|1434534580.186507|||'
- R=sip:25704386
Re: [SR-Users] usage of t_flush_flags() in event_route
Hi Daniel, thanks for the answer, that's what I was thinking - maybe the flags do not persist or are destroyed after the per-branch failure route. However, the t_flush_flags description says this function can be used in any route, so in should be fixed in the long term.. Let me check if I can come up with some workaround in config script for the time being. Regards, Andrew On 05/27/2015 08:19 AM, Daniel-Constantin Mierla wrote: Hello, I haven't used this event route and the flags, if they don't persist afterwards, then maybe it needs a patch to be fixed. Hugh implemented it, iirc, not sure if he was looking at this aspect. Cheers, Daniel ___ 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
[SR-Users] usage of t_flush_flags() in event_route
Hi Daniel and others,
I'm having a problem with acc module if I'm using the event_route/
branch-failure:
say, the call comes from the app server and goes to the registered user.
We arm the the failure route and per-branch failure route for the 302
redirect from the UA. We explicitly reset the accounting flags because
we don't want to account the calls from the app server. The transaction
is created implicitly by the t_relay().
Now if the UA responds with the 302 response and we are going to process
that, we want to create an acc record for the new target from 302
message because this call may incur additional costs. I'm setting the
accounting flags and calling t_flush_flags() but that doesn't work (no
accounting record). Any idea if I'm doing something wrong or maybe
there's a bug when changing the flags and then calling t_flush_flags
from the event_route?
Here are the module parameters:
modparam(acc, early_media, 0)
modparam(acc, report_ack, 0)
modparam(acc, report_cancels, 1)
modparam(acc, detect_direction, 1)
modparam(acc, db_flag, 1)
modparam(acc, db_missed_flag, 2)
modparam(acc, failed_transaction_flag, 3)
modparam(acc, db_url, PAIR_URL)
modparam(acc, db_extra,
src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd)
modparam(acc, multi_leg_info, src_leg=$avp(i:901);dst_leg=$avp(i:902))
modparam(acc, time_mode, 2)
modparam(acc, time_attr, time_hires)
modparam(acc, cdr_log_enable, 0)
FTR, we tried 4.1.6 and 4.1.8.
And here is the event route (with flags defined like this:
flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...):
event_route[tm:branch-failure:redirect]
{
route(ROUTE_STOP_RTPPROXY_BRANCH);
if($T_rpl($rs) == 301 || $T_rpl($rs) == 302)
{
# initialise variables when entering failure route
route(ROUTE_INITVARS);
# these need to be avps because we need it in reply/failure-route
$(avp(s:from_faxserver)[*]) = 0;
$(avp(s:to_faxserver)[*]) = 0;
$(avp(s:cf_from_pstn)[*]) = 0;
$(avp(s:from_pstn)[*]) = 0;
$(avp(s:proxylu_from_pstn)[*]) = 0;
$(avp(s:lcr_flags)[*]) = 0;
$(avp(s:em_call)[*]) = 0;
$(avp(s:from_pbx)[*]) = 0;
$(avp(s:p_to_device)[*]) = 0;
$(avp(s:p_to_group)[*]) = 0;
$(avp(s:is_primary)[*]) = 0;
# now let's process a 30x
$(avp(s:acc_state)[*]) = cfb;
$(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user);
$(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain);
$(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user);
$(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain);
$(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid);
$(avp(s:callee_uuid)[*]) = $null;
# the $var(no_acc) is 0 at this point but the flags may have
been reset
# if this is a call from PBX user - we do want accounting for
the 302 redirect
if(isflagset(FLAG_ACC_DB)) {
xlog(L_NOTICE, ++ ACC flag is set - [% logreq -%]\n);
} else {
xlog(L_NOTICE, -- ACC flag is NOT set - [% logreq
-%]\n);
}
setflag(FLAG_ACC_FAILED);
setflag(FLAG_ACC_DB);
t_flush_flags();
# get last URI from destination-set and set it as R-URI
$var(contact) = $T_rpl($ct);
$var(contact) = $(var(contact){nameaddr.uri});
if($var(contact) == 0 || $var(contact) == $null)
{
xlog(L_ERROR, Failed to fetch contact '$ct' from 301/302
- [% logreq -%]\n);
acc_db_request(480, acc);
$var(announce_handle) = callee_tmp_unavailable;
$var(announce_set) = $xavp(callee_real_prefs[0]=sound_set);
$(avp(s:announce_code)[*]) = 480;
$(avp(s:announce_reason)[*]) = Temporarily Unavailable;
route(ROUTE_EARLY_REJECT);
}
$ru = $var(contact);
xlog(L_NOTICE, Redirect from UAC intercepted - [% logreq -%]\n);
$(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU);
$(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td);
$var(forward) = 1;
$var(redirected_forward) = 1;
route(ROUTE_LOAD_CALLER_PREF);
route(ROUTE_FIND_CALLEE);
}
}
Thanks.
Andrew
___
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
Re: [SR-Users] Unix timestamp with millisecond precision
See http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#tv_name On 10/22/2014 10:40 AM, Grant Bagdasarian wrote: Hello, I’m currently using the $TS psuedovariable to get the current unix timestamp, but this only returns the timestamp up to a second precision. Is it possible in kamailio to get the current unix timestamp with milliseconds precision? Regards, Grant ___ 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
[SR-Users] textopsx issue when modifying same header several times
Hi,
we are filtering some method names from the Allow header with kamailio
4.1: depending in the configuration:
if (hf_value_exists(Allow, INFO))
{
xlog(L_INFO, Remove INFO from Allow\n);
exclude_hf_value(Allow, INFO);
}
if (hf_value_exists(Allow, REFER)) {
xlog(L_INFO, Remove REFER from Allow\n);
exclude_hf_value(Allow, REFER);
}
There is a problem in case multiple rewrites are done on the same header
so the resulting message is going to be broken:
Input:
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY,
PRACK, INFO, REFER
OUTPUT:
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY,
PRACK,
The trailing comma is not allowed. So we have to use
msg_apply_changes(). But kamailio is quite restrictive on where you can
call msg_apply_changes():
1. it must be done before record_route()* and
2. it is not allowed in the branch route. As for me it's valid use case
when you need to do some header manipulations only for some of the
branches, isn't?
*(1) is further aggravated by the fact that the record_route() for
loose-routed requests is usually called quite early in the processing
pipeline. Imagine kamailio.cfg would look like:
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# CANCEL processing
if (is_method(CANCEL))
{
if (t_check_trans()) {
route(RELAY);
}
exit;
}
# --- filter Allow
route(FILTER_ALLOW_METHODS);
# handle requests within SIP dialogs
route(WITHINDLG);
I don't like the idea of changing the message before loose-routing.
Taking (1) and (2) into account, would it be possible to
a) fix exclude_hf_value to allow to filter multiple values in a single
pass and
b) allow msg_apply_changes() in branch route without any ill effects on
the record-route headers?
Please share your ideas.
Thanks,
Andrew
___
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
Re: [SR-Users] howto synchronize ruid across multiple proxy instances
Hi, OK, thanks for the clarification. Maybe we will check the load_db_contacts(userid) way with Victor. Regards, Andrew On 09/15/2014 12:04 PM, Daniel-Constantin Mierla wrote: Hello, On 12/09/14 15:42, Andrew Pogrebennyk wrote: Hi, let's say I'm running two proxies/registrars that need to access Shared location DB in db_mode=1 (all changes to usrloc are immediately reflected in database too). I have observed that if the UAC re-registers before the previous registration's expiry and the new REGISTER reaches the other proxy than the one that processed the registration originally, this new proxy is going to insert the second record for the same username into location table instead of updating the existing registration. I assume the problem is the ruid calculation, if the proxy doesn't have the record in memory, it will not matter if it is with same ruid or not, it will create a new record. If it is the same ruid, it will be a failure when inserting in database, not updating it there, leaving the old record in place. Also, even if this one will overwrite in database, the other proxy will still have in memory and will route calls to it. Alex mentioned in a separate email the db only mode - that could be a solution. Also, you can try deleting the record from db via sqlops based on username, domain, contact uri if the registered() returns false for that user, before doing save(). If looking at writing C code to get it done, maybe it can be achieved with a new function to load_db_contacts(userid) to be called before save(). Back on ruid, just for sake of clarifications, by the way ruid is generated, is unlikely to get it the same across many systems, because it uses pid and local counters. Cheers, Daniel ___ 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
[SR-Users] howto synchronize ruid across multiple proxy instances
Hi, let's say I'm running two proxies/registrars that need to access Shared location DB in db_mode=1 (all changes to usrloc are immediately reflected in database too). I have observed that if the UAC re-registers before the previous registration's expiry and the new REGISTER reaches the other proxy than the one that processed the registration originally, this new proxy is going to insert the second record for the same username into location table instead of updating the existing registration. I assume the problem is the ruid calculation, because proxy2 generated a different ruid than proxy1 since internal state of the two nodes is not shared. I have two questions: 1) is that assumption correct? 2) is there any way to synchronize the ruid of the two nodes in usrloc module, similar to the secret parameter in the auth module, or a better solution? I prefer the load-balancer to select random proxy and not do a hash based on From URI. I don't have the trace of the two registrations at the moment, but could get one next week. The issue popped up when running some sipp tests, but it's also possible that some cheap SIP endpoint would register well before the expiration time, creating duplicate registration. So if someone dials its number it's going to receive multiple incoming calls. Here are the module parameters I am using, all quite ordinary: modparam(usrloc, use_domain, USE_DOMAIN) modparam(usrloc, db_mode, 1) modparam(usrloc, db_url, CENTRAL_URL) modparam(usrloc, db_check_update, 0) modparam(usrloc, nat_bflag, FLB_NATB) modparam(registrar, default_expires, 3600) modparam(registrar, min_expires, 60) modparam(registrar, max_expires, 43200) modparam(registrar, method_filtering, 0) modparam(registrar, append_branches, 1) modparam(registrar, max_contacts, 5) modparam(registrar, received_avp, $avp(s:received)) modparam(registrar, use_path, 1) modparam(registrar, path_mode, 0) modparam(registrar, path_use_received, 1) modparam(registrar, gruu_enabled, 0) Any ideas? Thanks in advance. Andrew ___ 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
Re: [SR-Users] no rtpengine.so module
On 03/07/14 16:42, Yuriy Gorlichenko wrote: thanks for fas reply. If I may user rtpengine as rtpproxy maybe you already use it or just know - does rtpengine provide ridge mode as rtpproxy between internal and external interfaces? At my instalne if I add rtpengine --ip=my.ext.net.addr/my.int.net.addr It shows that ip if wrong, but I neet proxy rtp between interfaces. Thanks. rtpengine does not support the bridge mode. Regards, Andrew ___ 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
Re: [SR-Users] Error with ACK from SIPML5
Sorry for reviving this old thread. I think I've come across the same issue as Jayesh and it is triggered by the Route inserted by sipml5 into in-dialog requests like ACK - Route: sip:172.18.101.48:5060;lr;sipml5-outbound;transport=udp Jayesh had two Route headers that belong to this kamailio instance and I also have two of them: Route: sip:172.18.101.48;transport=ws;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060 Route: sip:127.0.0.1;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060 Although RFC3261 doesn't seem to forbit that explicitly, but when sipml5 adds a pre-existing Route to *in-dialog* request, it creates a set where not all URIs are unique. What happens in kamailio is that when loose_route encounters the topmost Route from sipml5 client, it does not remove other routes before sending the request, e.g. I see no further processing after Next URI is a loose router. I can share the kamailio debug log, but the most important part is: Feb 14 23:26:54 spce lb[3773]: DEBUG: websocket [ws_frame.c:589]: ws_frame_receive(): Rx SIP message: ACK sip:ngcp-lb@172.18.101.48:5060;ngcpct=7369703a3132372e302e302e313a35303830 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPLN9elJsPlfeyLsMqN2J;rport From: 4252143385sip:4252143385@172.18.101.48;tag=RtycJXR6JIUFzAezAbRX To: sip:2067077495@172.18.101.48;tag=1F9894DE-52FE982E000437E0-321A4700 Contact: 4252143385sip:4252143385@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language=en,fr Call-ID: 1bff80ac-1e47-ab0f-3ea4-6a7b8637030b CSeq: 43084 ACK Content-Length: 0 Route: sip:172.18.101.48:5060;lr;sipml5-outbound;transport=udp Max-Forwards: 70 Proxy-Authorization: … Route: sip:172.18.101.48;transport=ws;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060 Route: sip:127.0.0.1;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060 Route: sip:127.0.0.1:5062;lr=on;ftag=RtycJXR6JIUFzAezAbRX;did=117.fcb;mpd=ii;ice_caller=replace;savp_caller=force_srtp;avpf_caller=force_avpf;ice_callee=strip;savp_callee=force_rtp;avpf_callee=force_avp;rtpprx=yes;vsf=ampNeU9nS3R6XXJyQ2QgJVJTaiZjfFJ4S35Wf0E- User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27 Organization: Doubango Telecom [...] Feb 14 23:26:54 spce lb[3773]: DEBUG: rr [loose.c:778]: after_loose(): Topmost route URI: 'sip:172.18.101.48:5060;lr;sipml5-outbound;transport=udp' is me Feb 14 23:26:54 spce lb[3773]: DEBUG: rr [loose.c:862]: after_loose(): URI to be processed: 'sip:172.18.101.48;transport=ws;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060' Feb 14 23:26:54 spce lb[3773]: DEBUG: rr [loose.c:871]: after_loose(): Next URI is a loose router Feb 14 23:26:54 spce lb[3773]: DEBUG: rr [rr_cb.c:97]: run_rr_callbacks(): callback id 0 entered with lr;sipml5-outbound;transport=udp Feb 14 23:26:54 spce lb[3773]: DEBUG: siputils [checks.c:106]: has_totag(): totag found Feb 14 23:26:54 spce lb[3773]: DEBUG: rr [loose.c:974]: check_route_param(): params are ;lr;sipml5-outbound;transport=udp Feb 14 23:26:54 spce lb[3773]: INFO: script: Reset loose-routing, du='sip:172.18.101.48;transport=ws;r2=on;lr=on;ftag=RtycJXR6JIUFzAezAbRX;nat=yes;ngcplb=yes;socket=ws:172.18.101.48:5060' - R=sip:127.0.0.1:5080 ID=1bff80ac-1e47-ab0f-3ea4-6a7b8637030b I see several people came to this mailing list with this issue, but I'm unsure about the outcome. Is this acceptable for an UAC to add a route to in-dialog ACK, which was not in the route set received from the UAS or proxy? Does it intend that the request should spiral? Did anyone take it to the sipml5.org guys? Is loose_route() behavior 100% valid? Thanks in advance. Andrew P.S. the log file from Jayesh is quoted below: On 24/06/13 17:11, Daniel-Constantin Mierla wrote: Can you get the ngrep on the server from the initial invite to the ack. It seems that record route is not properly mirrored, so one client might mess the route path. Cheers, Daniel On 6/23/13 8:58 PM, Jayesh Nambiar wrote: I did do debug=3 and saw the logs but couldn't figure out much. Here is the log for the appropriate ACK received on websocket port. I've highlighted a few lines I felt might be problematic: [...] Jun 23 18:45:46 v9 /usr/local/ghanti-ko/sbin/kamailio[22879]: DEBUG: rr [loose.c:778]: after_loose(): Topmost route URI: 'sip:126.128.68.9:5060;lr;sipml5-outbound;transport=udp' is me Jun 23 18:45:46 v9 /usr/local/ghanti-ko/sbin/kamailio[22879]: DEBUG: rr [loose.c:862]: after_loose(): URI to be processed: 'sip:126.128.68.9:8080;transport=ws;r2=on;lr=on;ftag=7EKISKAakFG2QZ2JvpDO;nat=yes' Jun 23 18:45:46 v9 /usr/local/ghanti-ko/sbin/kamailio[22879]: DEBUG: rr [loose.c:871]: after_loose(): Next URI is a loose router Jun 23 18:45:46 v9 /usr/local/ghanti-ko/sbin/kamailio[22879]: DEBUG: siputils [checks.c:106]:
Re: [SR-Users] reg event support in subscribe
Hi, See presence_reginfo - Extension to Presence server for registration info replication (RFC3680) http://kamailio.org/docs/modules/stable/modules/presence_reginfo.html On 02/05/2014 07:56 AM, Premchandiran wrote: Hi All, May I know whether kamailio supports event header with reg (EVENT:reg) (rfc 3680) , if kamailio supports may I know the module whether it is in presence or presence_xml or any other module? Regards, *Prem Chandiran M*** ___ 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
Re: [SR-Users] NAT helper frustrations with multiple kamailios
On 02/05/2014 11:20 AM, Daniel Tryba wrote: Off to figure out why uac_test incorrectly flags the onhold process as needing the nat helper... Well, nat_uac_test is pretty straightforward, it does what the flags tell it to do: http://kamailio.org/docs/modules/stable/modules/nathelper.html#idp1655008 What I do is skip nat_uac_test for the requests and responses coming from other proxies' IPs.. Andrew ___ 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
Re: [SR-Users] kamailio with mediaproxy-ng, 488 Not Acceptable Here
Hi, could you please post also your Chrome js developer log? Does the problem exist if you start the jssip clients without video support? Andrew On 02/03/2014 12:00 PM, Mihai Marin wrote: Hello, Another weekend struggling to make a call from jssip to another jssip behind firewall and I still receive 488 - Not Acceptable Here. I tried all the ideas that I had/received without any success - including catch 488 and re-invite. [...] What do I miss from my configuration? Thank you. Best regards, Mihai M ___ 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
Re: [SR-Users] Kamailio behind NAT
On 01/23/2014 05:12 PM, Klaus Darilion wrote: It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? If rtpproxy uses only a single listen-IP, then these flags are not needed. Only if you operate rtpproxy in bridge mode, then you need these flags. Bridge mode is necessary if you do not have IP routing between the internal network and the virtual external network, or if you want to bridge between IPv4 and IPv6. John, This function can be used to check the direction of every message: http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 You might also need to append the record-route parameters to remember the flags you have passed to the manage_rtpproxy() initially. Based on the direction of the request and initial flags you can determine what flags to use when calling manage_rtpproxy() for a given in-dialog requests and reply. Hope this helps. Andrew ___ 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
Re: [SR-Users] Update R-URI after lookup(location)
Hi, You should create a branch_route and perform this manipulation there.. BR, Andrew On 01/14/2014 10:53 AM, Igor Potjevlesch wrote: Hello, No one has an idea? I was thinking that each request goes to RELAY but even if I try to modify the R-URI in this route, it fails. I still had the contact URI taken from “location” instead of my modification. Regards, Igor. *De :*Igor Potjevlesch [mailto:igor.potjevle...@gmail.com] *Envoyé :* lundi 13 janvier 2014 17:15 *À :* sr-users@lists.sip-router.org *Objet :* Update R-URI after lookup(location) Hello, I do an association between X aliases with 1 contact. This contact is connected one time in the location. But sometimes, for some scenario, this contact can be connected 2 or 3 times and the INVITE are sent with parallel forking. When an INVITE is received, after lookup(“aliases”), I set the R-URI with the original SIP TO with the following instruction: avp_pushto($ruri/username, $tU); before relay. The issue occurred with the next contacts. They don’t passed through this instruction. How can I update all available R-URI in location before relay? Regards, Igor. http://www.avast.com/ Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection Antivirus avast! http://www.avast.com/ est active. ___ 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 ___ 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
Re: [SR-Users] Help with authenticating using Kamailio
Hi, check if this answers your question: http://kb.asipto.com/asterisk:index Andrew On 01/13/2014 04:19 PM, Kasinath wrote: Hi All, I just installed Kamailio in one server and Asterisk in another. Asterisk loads it sipusers info from database which is in Kamailio server. I don't know how to go further. How can I authenticate Asterisk users through Kamailio. I am trying to authenticate using a sipphone. But no luck. I am missing alot here. I know we can add users using the following command kamctl add username password But I already have users in asterisk realtime db. Is there any difference? Awaiting your reply, Thanks in advance, ___ 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
Re: [SR-Users] (no subject)
Hi, On 10/11/2013 08:14 PM, Fred Posner wrote: I use rwie and rwei flags but in ngcp-mediaproxy-ng e and i seems to be used for IPv4 / IPv6 .. ... I don't believe that mediaproxy-ng can be used to bridge two ipv4 networks; only bridging for ipv6 - ipv4. That's true, there's no bridge mode in mediaproxy-ng. Andrew ___ 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
Re: [SR-Users] [Spce-user] FYI: Package update(s) in release 2.8
Dear 2.8 users, this update includes the fixes for kamailio-lb config (multiple sockets-related): - added record-route param to store socket; review logic for relaying in-dialog requests (fallback to default send socket if it's not defined explicitly); fixed socket selection for ACK after 4xx and replies to in-dialog requests to callee; - also file descriptors limit has been lifted for sems. The upgrade is highly recommended if you run the 2.8 version. On 06/04/2013 12:16 PM, Sipwise Repository Tracker wrote: One or more updates have been released in NGCP release 2.8: * ngcp-templates-ce-asterisk was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-cdr-exporter was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-check-tools was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-cleanup-tools was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-hylafaxplus was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-iaxmodem was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-kamailio was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-lsb was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-mediator was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-mysql was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-nginx was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-odbc was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-reminder was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-sems was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-system was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-ce-vmnotify was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-asterisk was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-callingcard was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-cdr-exporter was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-check-tools was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-cleanup-tools was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-glusterfs was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-ha was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-heartbeat2 was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-hylafaxplus was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-hylafaxplus-diva was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-hylafaxplus-iax was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-iaxmodem was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-kamailio was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-lsb was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-mediaproxy-ng was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-mediator was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-monit was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-monitoring-tools was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-mysql was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-nginx was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-odbc was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-pushd was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-redis was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-reminder was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-sems was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-system was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-vmnotify was updated from version 2.8.6 to version 2.8.7 * ngcp-templates-pro-voisniff-ng was updated from version 2.8.6 to version 2.8.7 To apply the latest update(s) execute: apt-get update apt-get upgrade \ ngcp-update-db-schema ngcp-update-cfg-schema \ ngcpcfg apply If you encounter any problems please report a bug at the spce-u...@lists.sipwise.com mailing list. Further documentation is available at: http://www.sipwise.com/products/spce/documentation/ This is an automated mail by the Continuous Delivery platform at Sipwise.com ___ Spce-user mailing list spce-u...@lists.sipwise.com http://lists.sipwise.com/listinfo/spce-user ___ SIP Express Router (SER)
Re: [SR-Users] path uri problem
Hi Juha, On 04/07/2013 01:51 PM, Juha Heinanen wrote: i escaped them, but it didn't help. path header now looks like: Path: sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58156%3Btransport%3Dtcp'. and i still get the same error: Apr 7 14:49:47 wheezy1 /usr/sbin/sip-proxy[8709]: ERROR: registrar [save.c:887]: Failed to parse Path: URI I don't see why you think that ; and = should be escaped. rfc3327 chapter 4 says: The syntax for Path is defined as follows: Path = Path HCOLON path-value *( COMMA path-value ) path-value = name-addr *( SEMI rr-param ) Note that the Path header field values conform to the syntax of a Route element as defined in [1]. As suggested therein, such values MUST include the loose-routing indicator parameter ;lr for full compliance with [1]. The rules for Route element are as follows: Route= Route HCOLON route-param *(COMMA route-param) route-param = name-addr *( SEMI rr-param ) name-addr = [ display-name ] LAQUOT addr-spec RAQUOT addr-spec = SIP-URI / SIPS-URI / absoluteURI rr-param = generic-param generic-param = token [ EQUAL gen-value ] gen-value = token / host / quoted-string Why would someone want to escape semicolor (SEMI) which separated either Route or URI parameters? Also EQUAL used in pname=pvalue does not need escaping. We've already had a closer look at add_path_received() here at sipwise when we found double quotes in Route param value to be invalid and changed them to single ones. We have not observed the error in save() you have posted in kamailio 3.3.. Moreover, the ibc's Ragel-SIP-Parser suggests that the Path header above is correct :) So, from my PoV: - the ;transport=tcp;lr;received=... part are route-param's which follow the above rules. - the 'sip:192.98.102.10:58156;transport=tcp' part contains URI parameters. It is still fine according to definition of Path/Route element above - and agrees with the definition of other-param too: SIP-URI = sip: [ userinfo ] hostport uri-parameters [ headers ] uri-parameters = *( ; uri-parameter) uri-parameter = transport-param / user-param / method-param / ttl-param / maddr-param / lr-param / other-param other-param = pname [ = pvalue ] pvalue = 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = [ / ] / / / : / / + / $ unreserved = alphanum / mark mark = - / _ / . / ! / ~ / * / ' / ( / ) alphanum = ALPHA / DIGIT escaped = % HEXDIG HEXDIG Do you also have the save() problem in 3.3? Andrew ___ 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
Re: [SR-Users] path uri problem
On 04/09/2013 11:59 AM, Juha Heinanen wrote: because path-value starts with name-addr and my interpretation is that since there are s around this path header body: Path: sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58156%3Btransport%3Dtcp' solely consists of name-addr and does not include any rr-params. sip uri included in name-addr in turn cannot have ; and = in its param values. I see, probably you are right. We've had base64 encode/decode on our todo list for some time already (for broken UAs, e.g. some of them cut everything after '), so maybe we should offer user a choice of proper escaping and base64 encoding here. it turned out that save error had nothing to do with syntax of path header, but was due to a bug that i fixed. -- juha ___ 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
Re: [SR-Users] loose_route causing loop
Hello, Usually this happens when the RURI has the IP address/ domain which points to this server or which is present in the domain table. There are some clients which may put even server IP address into Contact of INVITE or 200 OK instead of its own so the further requests will be mis-routed. You should post the SIP trace and kamailio log with debug level 3 if this doesn't answer your question. On 31/01/2013 20:05, Renan Capaverde wrote: Hello, I have a problem with my configuration file. When I receive a in-dialog request (REINVITE) with topmost route being my kamailio server, it enters in loop and then the call is dropped. What is the easiest way to prevent this to happen? Best Regards, -- *Renan Capaverde* /Estagiário Nível Superior/ /DDT - STE - Soluções em Tecnologia e Embarcados/ *DÍGITRO TECNOLOGIA* *E-mail:* renan.capave...@digitro.com.br mailto:renan.capave...@digitro.com.br *Fone:* +55 48 3281-7000*Ramal:* 8132 *Fax:* +55 48 3281-7299 *Site:* www.digitro.com http://www.digitro.com /Antes de imprimir, pense na sua responsabilidade e no seu compromisso com o meio ambiente/ Esta mensagem, incluindo seus anexos, é reservada somente à Dígitro e ao destinatário da mensagem. Caso você tenha recebido esta mensagem por engano, queira por favor, retorná-la ao remetente e apagá-la de seus arquivos. ___ 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 ___ 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
Re: [SR-Users] REGISTER OK but on send 407 Proxy Authentication Required with no ACK
Carsten,
On 01/29/2013 08:25 PM, Carsten Maass wrote:
Or does it mean, the authentication is rejected because the local part
030123456789 in URI does not match the subscriber 979?
Indeed, this is what happens when you call auth_check with flag 1:
if (!auth_check($fd, subscriber, 1)) {..}
flags - set of flags to control the behaviour of the function. If it is
1, then the function will check to see if the authentication username
matches either To or From header username, a matter of whether it is for
a REGISTER request or not..
Glad you solved it.
Andrew
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] REGISTER OK but on send 407 Proxy Authentication Required with no ACK
Hello,
I see no problem with an ACK in the trace. The t38modem ACK's the 407,
sends the new INVITE with authorization but it's not accepted by
kamailio. I'm not sure if t38modem can even perform the authorization.
Maybe you need to accept the calls from it without authorization. But if
it can and you have configured the password in there, you should check
that the subscriber 979 exist in kamailio domain 10.1.1.148 and the
passwords do match.
On 01/29/2013 06:41 AM, Carsten Maass wrote:
Hi all,
I am trying to set up a fax gateway in the following way:
PSTN-GW (10.1.1.150) -- Kamailio (10.1.1.148:5123) -- t38modem
(10.1.1.148:6050) -- Hylafax
PSTN-GW is a standalone Berofix appliance and Kamailio 3.3.3, t38modem
1.2 and Hylafax 6.0.3 running on the same host under Redhat EL6.
I used the default kamailio.cfg and just adjusted the PSTN route pattern
to if(!($rU=~^(\+|00|0)[1-9][0-9]{3,20}$)), to route all calls
starting with 0 to the PSTN-GW.
Both PSTN-GW and t38modem Successfully register to Kamailio but when I
try to send out a fax from t38modem, Kamailio doesn't ACK the
Proxy-Authorization request and t38modem terminates the call:
...
/siptrace
What do I have to adjust to make this work?
Any help and pointers highly appreciated.
Thanks in advance and greetings,
Carsten.
___
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
Re: [SR-Users] LCR
Hello,
try printing the $ru with xlog statement when the request comes into the
server and in the beginning of route[LCR].
On 11/01/2013 01:50, Douglas Ugalde wrote:
Hi,
Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I
do to fix this error:
ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI
This is my LCR configuration block:
route[LCR] {
if(!load_gws(1)){
sl_send_reply(500, Internal server error, unable to load gateways);
xlog(L_NOTICE,Internal server error, unable to load gateways);
break;
}
if(!next_gw()){
sl_send_reply(503, Service not available, no gateways found);
break;
}
}
I not sure if this configuration thats ok, please somebody help me.
Note:
Params and modules are already loaded, sorry for my english.
Best regards.
___
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
[SR-Users] why use received as RURI for nathelper OPTIONS ping?
Some Fritzbox devices don't accept the OPTIONS ping generated by
kamailio with RURI: sip:11.22.3.4:5060. In the location table we have:
received: sip:11.22.3.4:5060
contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967
path: sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'
Kamailio version is 3.3.2 and the nathelper config is nothing fancy:
modparam(nathelper, natping_interval, 15)
modparam(nathelper, ping_nated_only, 1)
modparam(nathelper, sipping_bflag, FLB_NATSIPPING)
modparam(nathelper, sipping_from, sip:pinger@sipwise.local)
modparam(nathelper, received_avp, $avp(s:received))
What I don't understand is why kamailio sets RURI of the OPTIONS message
to value of received instead of the contact. I suspect a bug in the
parser somewhere along these lines:
rval =
ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0),
((unsigned int)(unsigned
long)timer_idx)*natping_interval+iteration,
natping_processes*natping_interval);
if (rval != 0) {
pkg_free(buf);
goto done;
}
}
if (buf == NULL)
goto done;
cp = buf;
while (1) {
memcpy((c.len), cp, sizeof(c.len));
if (c.len == 0)
break;
c.s = (char*)cp + sizeof(c.len);
cp = (char*)cp + sizeof(c.len) + c.len;
memcpy( send_sock, cp, sizeof(send_sock));
cp = (char*)cp + sizeof(send_sock);
memcpy( flags, cp, sizeof(flags));
cp = (char*)cp + sizeof(flags);
memcpy( (path.len), cp, sizeof(path.len));
path.s = path.len ? ((char*)cp + sizeof(path.len)) : NULL ;
cp = (char*)cp + sizeof(path.len) + path.len;
/* determin the destination */
if ( path.len (flagssipping_flag)!=0 ) {
/* send to first URI in path */
if (get_path_dst_uri( path, opt) 0) {
LM_ERR(failed to get dst_uri for Path\n);
continue;
}
/* send to the contact/received */
if (parse_uri(opt.s, opt.len, curi) 0) {
LM_ERR(can't parse contact dst_uri\n);
continue;
}
} else {
___
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
Re: [SR-Users] why use received as RURI for nathelper OPTIONS ping?
On 01/07/2013 01:29 PM, Andrew Pogrebennyk wrote: What I don't understand is why kamailio sets RURI of the OPTIONS message to value of received instead of the contact. I suspect a bug in the parser somewhere along these lines: rval = ul.get_all_ucontacts(buf,cblen,(ping_nated_only?ul.nat_flag:0), This needs some overhaul. The get_all_mem_ucontacts prefers received over contact. So what nathelper does is set Path as $du and Received as $ru, then send it. But in case home proxy which generated the NAT ping is sitting behind the edge proxy and the user is behind NAT, we need to pass both Contact and Received to the edge proxy. It looks like we (Sipwise) need to introduce a few modparams so the user choose what to put into $ru and $du (like 1 - Contact, 2 - Received, 3 - Path). I'm just wondering if there is anything speaking against that or missing in the light of SIP-Outbound implementation. ___ 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
[SR-Users] maximum number of branches exceeded after kamalilio restart
Hello, I'm getting some errors - if I restart kamailio on a live system, the first call after restart fails with max nr of branches exceeded. However, we are not doing append_branch() or anything fancy for that call at all: Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: core [dset.c:306]: max nr of branches exceeded Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: registrar [lookup.c:298]: failed to append a branch Also it appears on the other system with a different config: Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:656]: ERROR: add_uac: maximum number of branches exceeded Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed The second call however goes through and this error does not appear until the next kamailio restart. What could be the problem here? P.S. The kamailio version is 3.3.2. Andrew ___ 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
Re: [SR-Users] Difficulties with with loose_route() when trying to handle non-compliant ACK
Richard, well, the RURI of remote ACK has proxy IP address 10.200.70.100 so proxy thinks that previous hop was a strict router. I can't think of any workaround that would not be an ugly hack at the moment, though. On 11/23/2012 03:24 AM, Richard Brady wrote: Hi guys I have a multihomed Kamailio proxy sitting between two B2BUAs on separate networks and record-routing all dialogs. The problem I have is that when one of these devices receives a 200 OK, it does not populate the RURI of the ACK correctly. Instead of taking it from the Contact header on the 200 OK, it uses the user part from the Contact header and sets the domain to the proxy IP. It then also populates the Route headers. The ACKs are below and the IPs are: 10.152.1.92:5060: UAC on outside 10.200.70.100:5060: proxy outside interface 192.168.242.100: proxy inside interface 192.168.242.102: UAS on inside This the ACK message going into and coming out of the proxy. # U 10.152.1.92:5060 - 10.200.70.100:5060 ACK sip:natted_ua*9197**192.168.242.102*5080*udp@10.200.70.100 SIP/2.0. Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK-2280-1-5. Route: sip:10.200.70.100;r2=on;lr=on;nat=yes,sip:192.168.242.100;r2=on;lr=on;nat=yes. From: sipp sip:sipp@127.0.1.1:5060;tag=2280SIPpTag001. To: sut sip:9197@10.200.70.100:5060;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280@127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.200.70.100:5060 - 192.168.242.100:5060 ACK sip:192.168.242.100;r2=on;lr=on;nat=yes SIP/2.0. Via: SIP/2.0/UDP 10.200.70.100;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=10.152.1.92;branch=z9hG4bK-2280-1-5. Route: . From: sipp sip:sipp@127.0.1.1:5060;tag=2280SIPpTag001. To: sut sip:9197@10.200.70.100:5060;tag=p0FaeQ1QUS9ae. Call-ID: 1-2280@127.0.1.1. CSeq: 1 ACK. Contact: sip:sipp@127.0.1.1:5060;alias=10.152.1.92~5060~1. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. So I am trying to understand why it is trying to relay to itself (I have mhomed=1) and why it is rewriting the RURI as if it is a strict router. Currently decode_contact() is disabled but enabling doesn't seem to help. Unless there is a very specific place where it belongs. Any advice would be hugely appreciated. I can always paste logs / configs / traces. Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbr...@gmail.com ___ 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 ___ 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
Re: [SR-Users] Supporting TLS and DTLS in RTP Proxy
Kamal, perhaps RFC 5763 provides you some of the answers? On 10/16/2012 11:06 AM, Kamal Palei wrote: Hi Johansson, All Sincier regards and thanks for input. As I understand, all media packets pass through RTP Proxy. The RTP Proxy will receive simple UDP media packets from endpoints. Next RTP proxy today pass those RTP packets to destination party. My job is precisely to support TLS and DTLS path between RTP Proxy and destination party. In my setup the destination party is a media server. Do you really see a risk to have this setup. If so, please elaborate. I underdstand here the challenge setup TLS/DTLS connection with media server and send/recv media packets with server usuing either TLS or DTLS. Also you mentioned There's also solutions for RTP over DTLS , can you please share from where I can get the reference solution, it help me to great extent. Best Regards Kamal On Tue, Oct 16, 2012 at 12:08 AM, Olle E. Johansson o...@edvina.net mailto:o...@edvina.net wrote: 15 okt 2012 kl. 13:24 skrev Peter Lemenkov lemen...@gmail.com mailto:lemen...@gmail.com: Hello. 2012/10/15 Kamal Palei palei.ka...@gmail.com mailto:palei.ka...@gmail.com: Hi All I am planning to enhance RTP proxy to support TLS and DTLS. We have some requirements where we need to send RTP packets either over TLS or over DTLS. Shouldn't it be better to rely on SRTP/ZRTP instead rather than making your own incompatible realisation? SRTP use DTLS for key exchange. There's also solutions for RTP over DTLS, but the recommended way is DTLS+SRTP. This is what's standardized for WebRTC, and the way forward for SIP media as well. However, I don't see how RTPproxy can be the endpoint for DTLS key exchange, since it breaks the end2end path. Clients should use TURN relays... Curious on how you see this working! /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ 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 ___ 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
Re: [SR-Users] DB accounting missing if t_newtran() is called explicitly
Hi Daniel, Thank you for this advice.. I'm still struggling to get it work, still no luck even with t_flush_flags() immediately after setflag(). Maybe I will try to reproduce it during the weekend with kamailio default config. FTR the version used is kamailio 3.3.1. On 09/27/2012 05:17 PM, Daniel-Constantin Mierla wrote: Hello, you can try after you set the flag you wanted to be in transaction, to be sure it gets there. Cheers, Daniel On 9/26/12 7:43 PM, Andrew Pogrebennyk wrote: Hi Daniel, No, I don't. Thanks for the tip. Could you advice where t_flush_flags() should be placed? I tried in branch_route and immediately before t_relay(), it didn't help.. On 09/26/2012 05:53 PM, Daniel-Constantin Mierla wrote: Hello, do you use t_flush_flags()? http://kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2543767 Cheers, Daniel On 9/26/12 3:30 PM, Andrew Pogrebennyk wrote: Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply(100, Trying); xlog(L_INFO, New request - $ci\n); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life. ___ 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 ___ 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
[SR-Users] DB accounting missing if t_newtran() is called explicitly
Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply(100, Trying); xlog(L_INFO, New request - $ci\n); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life. ___ 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
Re: [SR-Users] DB accounting missing if t_newtran() is called explicitly
Hi Daniel, No, I don't. Thanks for the tip. Could you advice where t_flush_flags() should be placed? I tried in branch_route and immediately before t_relay(), it didn't help.. On 09/26/2012 05:53 PM, Daniel-Constantin Mierla wrote: Hello, do you use t_flush_flags()? http://kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2543767 Cheers, Daniel On 9/26/12 3:30 PM, Andrew Pogrebennyk wrote: Hi, I have found recently that in order to detect retransmits I have to create a transaction explicitly when the request comes in: force_rport(); if(!t_check_trans()) t_newtran(); sl_send_reply(100, Trying); xlog(L_INFO, New request - $ci\n); it appears like there are carriers or UAs that do not honor the T1 retransmission interval retransmit the INVITE sooner than proxy creates a transaction in t_relay(). And since we are counting concurrent calls, we count the same call multiple times, which is not good. But with this patch we've faced another sporadic problem - if the transaction is created beforehand the accounting record is lost.. we use acc_db mode and set flag to account the transaction. And there are no errors in kamailio log but no insert into acc in mysql binlog either. I wasn't successful reproducing it in the lab systems with identical setup. Is anybody here perhaps aware of some limitation in acc module or callbacks which makes a transaction created beforehand not accountable? On a related note, it could make sense to create a transaction implicitly if dlg_manage() is called to avoid counting same call many times, I just don't know yet how common this issue is in real life. ___ 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 ___ 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
Re: [SR-Users] IM on Kamailio
On 09/18/2012 12:15 PM, Gary Shergill wrote: Note that I am testing this with one computer connected by Bria and another computer connected via Blink. I am able to log on to a user on each (test1 and test2) and they are able to call each other. The issue is, with presence enabled, they are unable to IM each other (or add each other as contacts and see online status). Gary, I thought Bria uses RPID data format for presence (RFC 4480) while Blink uses PIDF so they won't be able to see presence status of each other. I see though that blink website mentions RPID as well, maybe somebody more knowledgeable about blink can correct me. For IM, add MESSAGE method to supported methods and send it after lookup like INVITE. For offline message delivery, checkout the msilo module readme. HTH, Andrew ___ 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
Re: [SR-Users] TCP Path received quotes
Hello Spencer, actually double quotes are not allowed in URI parameter. In the BNF grammar the allowed chars in the unreserved definition are alphanum and mark, where mark is only - / _ / . / ! / ~ / * / ' / ( / ) ). This is already fixed in 3.3.0 if I am not mistaken, please check http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=a5b181bca6bf37b4a18ef502717b50d06e53d5e4 - maybe that helps you with your FreeSwitch. On 09/15/2012 10:51 AM, Spencer Thomason wrote: I see. FS complains about no transport protocol and gives a 503 with a header like that. Shouldn't they default to UDP in the absence of a transport parameter? It seems they are not honoring the quotes. In this setup Kamailio handles NAT traversal and forwards the registers to Freeswitch. This works: Path: sip:a.a.a.a:5070;lr;received=sip:b.b.b.b:5185;transport=tcp;transport=udp This does not: Path: sip:a.a.a.a:5070;lr;received=sip:b.b.b.b:5185;transport=tcp Spencer ___ 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
Re: [SR-Users] Best tool for crafting sip messages
Same here, also you can use protoshoot provided by kamailio, see utils/protoshoot. On 08/13/2012 07:45 AM, Mark Anthony Delfin wrote: Hi Anton, Previously I used the following. sipsak http://sipsak.org/ or sipp http://sipp.sourceforge.net/ Regards, Mark ___ 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
Re: [SR-Users] Remove m=video from SDP header
You might need to upgrade to 3.3.1 to use sdpops module:
http://kamailio.org/docs/modules/stable/modules/sdpops.html
On 08/13/2012 09:52 AM, phillman25 wrote:
Dear List
I am trying to remove specific lines from the following original SDP body:
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11602 RTP/AVP 34 99
a=rtpmap:34 H263/9
a=rtpmap:99 H264/9
a=sendrecv
I need to remove the following lines from the above SDP body as my
International carrier does not support Video capabilities:
m=video 15042 RTP/AVP 34 99
a=rtpmap:34 H263/9
a=rtpmap:99 H264/9
I used the following code in Kamailio config:
if(has_body(application/sdp) search_body(m=video)){
subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #');
subst_body('#a=rtpmap:34 (.*)$# #');
subst_body('#a=rtpmap:99 (.*)$# #');
subst_body('#a=sendrecv(.*)$# #');
}
Content-Type: application/sdp
Content-Length: 325
P-hint: outbound
v=0
o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
b=CT:384
t=0 0
m=audio 11904 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
As you can see the m=video body has been removed, however, calls are
still failing. Is there something i have missed?
I am using Kamailio v3.2.2
Thanking you in advance!
Phillip
___
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
___
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
Re: [SR-Users] 302 redirect with 2 numbers registered
Mino, I am not sure, but you could try the following:
set failure_reply_mode 3
(http://kamailio.org/docs/modules/stable/modules/tm.html#failure_reply_mode),
then handle 302 redirect in the proxy and use the contact as a new
branch like this:
if(status == 302)
{
$var(contact) = $ct;
$var(contact) = $(var(contact){nameaddr.uri});
$du = $var(contact);
append_branch();
t_relay();
}
Would it ring it while branch to Phone1 is still active? Maybe not, but
you will need to try..
On 08/03/2012 03:53 PM, Mino Haluz wrote:
Hi,
one number is registered on 2 phones. Phone1 has Always redirect set
to another number. When incoming call is initiated, Phone2 is ringing
and Phone1 sends 302 to the proxy. However the proxy does not send 302
to the caller (for ex. GW), but it waits for timeout of the Phone2.
Then the proxy sends 302 to the caller.
Can I do in kamailio, that it will ring on the Phone1 and also on the
number where it is redirected? I know kamailio is a proxy and cannot
initiate a call, but is there any solution? Thanks.
Mino
___
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
___
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
[SR-Users] New Wiki Site has problems?
At least some links at http://www.kamailio.org/wiki/ don't work at the moment: Install Kamailio v3.3.x From GIT Upgrade Kamailio v3.1.x to v3.2.0 Upgrade Kamailio v3.2.x to v3.3.0 you are redirected to some instruction from DokuWiki Installer when trying to visit them. Could somebody please check it? Thanks. Andrew ___ 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
Re: [SR-Users] lcr.dump_gws problem
Hi Gary, It was fixed already by Richard in 3.3 branch: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=28be16549831df46dd1b8312da223b02359d8a9c (and master) Thank you for the report. On 07/24/2012 09:44 PM, Gary Chen wrote: Sorry, it should be Kamailio 3.3.0 not 3.2.0. *From:*sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of *Gary Chen *Sent:* Tuesday, July 24, 2012 3:39 PM *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Subject:* [SR-Users] lcr.dump_gws problem Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary ___ 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 ___ 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
Re: [SR-Users] lcr.dump_gws problem
Hi Gary, Yes, that's an old/known issue. You can convert the IP to common format with a simple command: perl -MSocket -le 'print inet_ntoa(pack(N, 1273060816))' Maybe somebody from the kamailio team will take time to fix this. On 07/24/2012 09:39 PM, Gary Chen wrote: Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary ___ 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
Re: [SR-Users] kamailio core request dropping - no reply
On 06/19/2012 02:06 PM, Uri Shacked wrote: I am testing kamailio replies when an INVITE or another request arrives with lets say, VIA header missing The core drops the request. But, there is no reply for the originator (so it keep on resending the request...) Why? If there was no Via header, the recipient would have no way to know where to send the response. ___ 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
Re: [SR-Users] Looking for RTP Proxy in TCP
The papers talk about transport protocol for signaling, not media/RTP. I didn't hear of anyone who does RTP over TCP neither. I doubt even that the performance is a primary reason behind that, for media over TCP the client link must be virtually packet-loss free (due to TCP retransmissions), while over UDP sometimes up to 5% packet loss can be tolerated. TCP was not designed as transport for real-time media :-) On 06/08/2012 12:35 AM, Yang Hong wrote: Hello. SIP over TCP would reduce server performance significantly when compared with SIP Over UDP. Please read the following two papers. Combining RTP proxy with SIP over TCP would degrade SIP server performance even worse. --- http://www.cs.columbia.edu/~hgs/papers/Shen1008_TLS.pdf The Impact of TLS on SIP Server Performance Securing SIP is accomplished by using TLS instead of UDP as the transport protocol. We show that using TLS can reduce performance by up to a factor of 17 compared to the typical case of SIP-over-UDP. Network operators considering deploying SIP over TLS will need to consider the extra resources required to provide the same service quality as would be the case with UDP. --- http://www.cs.columbia.edu/~hgs/nossdav/2007/files/file-27-session5-paper1-nahum.pdf Evaluating SIP Proxy Server Performance The next most signi cant performance feature is which transport protocol is used, TCP or UDP. Using TCP can reduce performance anywhere from 43 percent (the stateful proxying scenario with authentication) to 65 percent (state-less proxying without authentication). --- Best regards, Yang Date: Thu, 7 Jun 2012 13:36:39 +0200 From: mico...@gmail.com To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Looking for RTP Proxy in TCP Hello, On 6/4/12 7:14 PM, Austin Einter wrote: Hi All Now I am using Kamailio 3.1.5 and RTP proxy 1.1. Looks both are compatible and working fine. The RTP Proxy basically sends/receives RTP packets over UDP. Is there any RTP Proxy available that does send/receive of RTP packets over TCP and also should be compatible with Kamailio 3.1.5. If you have any information in this regard, kindly share. RTP itself is specified over UDP, also I am not aware of any SIP phone doing RTP over TCP. MSRP is a mechanism specified for sending message streams over TCP, we have a module for that, but I guess is not exactly what you are looking for: http://kamailio.org/docs/modules/devel/modules/msrp.html Maybe based on it you can implement one that fits your needs. Cheers, Daniel ___ 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
Re: [SR-Users] [NOT Solved]Creating RURI ($ru) from Contact ($ct)
Hi, On 05/30/2012 12:22 PM, Aft nix wrote: So i'm interested if RFC 3261 provides any mechanism by which we can differentiate a BYE whether its from caller or callee. Check out is_direction() function: http://www.kamailio.org/docs/modules/3.2.x/modules_k/rr.html#id2527009 ___ 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
Re: [SR-Users] Config include
Hello, It is already there, see http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x On 05/09/2012 06:04 PM, Konstantin M. wrote: Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include path/to/some/config.conf). Is it possible to implement a such feature ? Thanks! ___ 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
Re: [SR-Users] Config include
Konstantin, You should put the include_file directive after loadmodule and modparam directives. So it can be either before main route block or at the bottom of your main kamailio.cfg. On 05/09/2012 06:48 PM, Konstantin M. wrote: After including a part of main config to included file -- I got a several errors like: 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command is_method 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 4, column 55: unknown command, missing loadmodule? 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command xlog 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 9, column 101: unknown command, missing loadmodule? ___ 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
Re: [SR-Users] reply code for lower cseq
Hi Daniel, See RFC 3261 section 12.2.2: If the remote sequence number was not empty, but the sequence number of the request is lower than the remote sequence number, the request is out of order and MUST be rejected with a 500 (Server Internal Error) response. However, 400 or some 4xx response would seem more reasonable to me, to let the UAC know it just did something wrong. And I'm not the only one: http://comments.gmane.org/gmane.ietf.sip-implementors/8970 On 04/24/2012 02:10 PM, Daniel-Constantin Mierla wrote: Hello, I was wondering if someone here can point quickly where specs mention what is the right reply code to send when a request within dialog is received with lower cseq value than the previous request. I couldn't spot the part in the RFC yet, if any related exists. Cheers, Daniel ___ 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
Re: [SR-Users] in doubt about in-dialog INVITE Route params processing
Hi, despite my initial doubt it works well :-) Thank you Daniel. On 04/04/2012 05:51 PM, Daniel-Constantin Mierla wrote: Hello, is record_route() executed as well as add_rr_param() for reinvites? Cannot spot in the logs. You can load debugger module and enable cfgtrace to see what actions are executed from the config file. Cheers, Daniel ___ 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
[SR-Users] in doubt about in-dialog INVITE Route params processing
I've been pondering an issue with Route header parameters being not mirrored by kamailio proxy into Record-Route field on in-dialog requests over the last few hours so I thought I'd just whether I'm missing something obvious. The call scenario in UA - lb - proxy - sems. Everything is fine with initial INVITE transaction where we add rr param ;rtpprx=yes. (We also use dialog module.) The UA does re-INVITE where it mirrors RR headers to Route set just fine. However, when forwarding re-INVITE kamailio proxy removes own Route header and inserts RR header without rtpproxy, did etc params. The kamailio default config from branch 3.1 routes in-dialog requests in the same fashion and I can't tell really if this is how it is supposed to work? I can workaround this by add_rr_param() but shouldn't missing did param cause a problem with dialog matching? Here is the debug with log level 3: Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:127.0.0.1:5062;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-' is me Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: core [parser/msg_parser.c:103]: found end of header Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:257]: No next Route HF found Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:811]: No next URI found Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [rr_cb.c:97]: callback id 1 entered with lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M- Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_handlers.c:910]: route param is '8a3.aab32831' (len=12) Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:442]: ref dlg 0x7f0a0be918b0 with 1 - 3 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:444]: dialog id=327302058 found on entry 936 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:757]: dialog 0x7f0a0be918b0 changed from state 4 to state 4, due event 8 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_handlers.c:1042]: sequential request successfully processed Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_timer.c:117]: inserting 0x7f0a0be91900 for 127601858 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:411]: cseq is 10760 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [rr_cb.c:97]: callback id 0 entered with lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M- Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:421]: getting 'vsf' Route param Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:429]: route param is 'aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-' (len=44) Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:479]: decoded uris are: new=[sip:43991001@192.168.51.210] old=[sip:sipwise-user1@192.168.51.210] Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are ;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M- Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: siputils [checks.c:76]: totag found Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: textops [textops.c:1789]: content type is 196611 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are ;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M- Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are ;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M- Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: INFO: script: Use mediaproxy for forward direction for IPv4/IPv4 - M=INVITE R=sip:127.0.0.1:5080 F=sip:sipwise-user1@192.168.51.210 T=sip:43991002@192.168.51.210 IP=192.168.51.1:35189 (127.0.0.1:5060) ID=M6obxlxpUOPPR3CQYE-d2t9-fgybDo6y Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rtpproxy [rtpproxy.c:2260]: proxy reply: 30080 192.168.51.210 4#012 Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: INFO: script: Relaying request - M=INVITE R=sip:127.0.0.1:5080 F=sip:sipwise-user1@192.168.51.210 T=sip:43991002@192.168.51.210 IP=192.168.51.1:35189 (127.0.0.1:5060) ID=M6obxlxpUOPPR3CQYE-d2t9-fgybDo6y ___ 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
Re: [SR-Users] Kamailio modifying ruri during loose-route
Dan, Well, it looks like kamailio recognizes 127.0.0.1 as its own URI that's why it does rewrite. Do you have by chance such alias in your config (or auto_aliases=yes)? On 03/26/2012 03:47 PM, Dan-Cristian Bogos wrote: Hey Guys, I have noticed some unexpected behavior (at least by me) during my tests. When routing specific ACK message back to 200 OK originator, Kamailio will rewrite the ruri with the value of route header. Is that known or some memory access problem? Call setup: B2BUA(127.0.0.1) - Kamailio (1.2.3.4) - End device (2.3.4.5). Bellow is a trace of 200 OK -ACK flow before and after kamailio, together with the xlog capture of ruri before and after calling loose_route on ACK. Thanks in advance for any kind of tip. DanB ___ 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
Re: [SR-Users] OT: is it allowed to send requests without waiting for the response of the previous request?
Hello Klaus, please check this thread: http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg10537.html On 03/14/2012 05:11 PM, Klaus Darilion wrote: I wonder if a presence server may send a NOTIFY if the previous NOTIFY in the dialog did not received an answer yet. I greped the relevant RFCs but couldn't find a definition. Thus, pointers are appreciated. HTH. Andrew ___ 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
Re: [SR-Users] Rewrite $tU
On 03/07/2012 08:46 PM, Lucas Alvarez wrote:.
Something like this:
if(($rU=~^(box02)[0-9]{2,15}$)) {
$rU = $(rU{s.substr,5,0});
$ru = sip: + $rU + @ + $sel(cfg_get.box02.gw_ip) + :
+ $sel(cfg_get.box02.gw_port);
}
BTW alternatively you can use dialplan the module to hold both regexp to
match and target IP with added benefit of in-database provisioning.
The problem I'm having is I'm not being able to do blind tranfers. I
think the cause is the prefix that remains in the TO field. After
rewriting the TO field nothing change. I would appreciate if someone
could point me to the right path.
Hmm, the blind transfers should be transparent to the proxy. So the
REFER request should go all the way through box02 to box01 and to the
caller, which should send a new INVITE so the box01 can apply the same
logic of finding out where the subscriber is registered to RURI.
What is the actual call flow that you see after REFER?
___
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
Re: [SR-Users] Rewrite $tU
On 03/07/2012 07:37 PM, Lucas Alvarez wrote: I want rewrite $tU but I'm not being able, I'm doing the following: remove_hf(To); insert_hf(To: sip:$rU@$rd\r\n, From); Then I'm printing $tU and it is still having the previous value, any help will be appreciated. Thanks in advance. Did you verify what is actually sent on the wire? If it's just a logging issue you need to do msg_apply_changes(). ___ 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
[SR-Users] UAC canceling a specific early dialog with BYE
Hello, we have a Kamailio proxy which gets the call from PSTN gw and does some call forwarding (serial forking) to several destinations through our sbc The call flow I am looking at is: - Kamailio sends INVITE to branch_1. - branch_1 sends 180 with to-tag*, proxy relays it to the gw * 180 meets the requirements for dialog creating 18x responses in sections 12.1, 12.1.1 because it contains to-tag, contact and mirrored record-route. - After some seconds Kamailio sends a CANCEL to branch_1. - And sends the INVITE to branch_2. - branch_1 replies 200 for the CANCEL and 487 for the INVITE. - branch_2 replies 180 and 200 for the INVITE. - When PSTN gw receives that it sees it still needs to cancel other early dialog established by 180 from branch_1. - The PSTN gw sends a BYE with to-tag of branch_1 to cancel this specific early dialog. SIP allows early dialogs to individually released while other dialogs continue, as written in RFC, section 15: The BYE request is used to terminate a specific session or attempted session. In this case, the specific session is the one with the peer UA on the other side of the dialog. (...). The caller’s UA MAY send a BYE for either confirmed or early dialogs, and the callee’s UA MAY send a BYE on confirmed dialogs, but MUST NOT send a BYE on early dialogs. The BYE follows the loose routing path, proxy gets 481 from the sbc and forwards that response back to PSTN gw, which somehow breaks it. AFAICS it's not specified in RFC what should the behavior look like when getting both a 200 and error-class response for the INVITE (quotes are most welcome!). IMO it would be more correct to absorb BYE in proxy but I see a big problem here: branch_2 can even ring for 5 minutes and it's not feasible for proxy to have a wt-timer that long. Also it's not possible to inform the gw that early dialog has cleared as soon as we receive 200/487 from branch_1. So I'm not sure which party is at fault and if we can workaround that somehow in the Kamailio. Any thoughts? ___ 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
[SR-Users] image missing in xmlrpc module doc
Hi, Was reading through the xmlrpc docs, could anybody with an access check why Figure RPC Example is missing from http://kamailio.org/docs/modules/devel/modules/xmlrpc.html#fig.rpc_example as well as 3.1 and 3.2 module docs? Thanks, Andrew ___ 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
Re: [SR-Users] ndb_redis module fails after a while
Hi Daniel, On 02/17/2012 10:47 AM, Daniel-Constantin Mierla wrote: I made a patch for server reconnect -- I had no access to a computer with redis lib installed for the moment, hopefully it compiles. If you can try and tell the result, it would be great, I can commit then. I may be able to test this patch as well. Currently compilations bails out on attempt to redeclare redisc_reconnect_server function parameter: CC (gcc) [M ndb_redis.so] ndb_redis_mod.o CC (gcc) [M ndb_redis.so] redis_client.o redis_client.c: In function ‘redisc_reconnect_server’: redis_client.c:206:19: error: ‘rsrv’ redeclared as different kind of symbol redis_client.c:202:46: note: previous definition of ‘rsrv’ was here make[1]: *** [redis_client.o] Error 1 make: *** [modules] Error 1 ___ 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
Re: [SR-Users] redirect client to keep original ruri
Since this thread will probably end up in Google I''ll share my
experience. I ended up with this
if(t_check_status(301|302))
{
#NOTE: must assign to $du to keep R-URI intact
$var(contact) = $T_rpl($ct);
$var(contact) = $(var(contact){nameaddr.uri});
$du = $var(contact);
xlog(L_INFO, Redirect from proxy intercepted - M=$rm
R=$ru F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n);
append_branch();
route(ROUTE_RELAY);
exit;
}
where route(ROUTE_RELAY) is merely a call t_relay_to(0x01) wrapped up
in some logging. So far so good.
There was a problem when a new request target needed digest-ch'd the
caller with 407. The redirect server by default relayed not 407 but 302
reply back. I've set tm modparam failure_reply_mode to 3 - voila!
http://kamailio.org/docs/modules/3.1.x/modules/tm.html#failure_reply_mode
___
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
Re: [SR-Users] Kamailio with Private IP - ACK for 200 OK problem
On 01/26/2012 08:21 PM, Krishna Kurapati wrote: Is there a configuration option to let kamailio use Public IP when setting record-route in 200 OK? Of course, you need record_route_preset() - see http://kamailio.org/docs/modules/3.1.x/modules_k/rr.html#id2667590 Also I would expect you need to set advertised_address core parameter if it's not set yet.. ___ 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
[SR-Users] redirect client to keep original ruri
Hello, assuming that I want to use contact from 302 response as outbound proxy but keep the original Request-URI, what should I do? Calling the revert_uri() after get_redirects() in failure_route doesn't do the trick. ___ 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
Re: [SR-Users] redirect client to keep original ruri
Daniel, thank you - method (2) works like a charm. On 01/25/2012 03:16 PM, Daniel-Constantin Mierla wrote: an option - get the contact header from 302 in failure route via $T_rpl($ct). Use its uri to set $du. I was getting an error for some reason: 0(7860) ERROR: core [pvapi.c:516]: error searching pvar T_rpl 0(7860) ERROR: core [pvapi.c:720]: wrong char [$/36] in [$T_rpl($ct)] at [7 (5)] version: kamailio 3.1.5 Alternative, set onreply_route and if it is 302 reply, take the contact uri and add it in an avp. Use that avp in failure route. This works. Regards, Andrew ___ 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
Re: [SR-Users] redirect client to keep original ruri
On 01/25/2012 03:18 PM, Alex Balashov wrote: Put the 302 Contact URI in the destination set instead, i.e. $du, not $ru. That will cause it to be relayed to the redirect destination on the network and transport level, but the logical target will remain the same. Ok. I was thinking that uac_redirect is the way to go, but it seems there isn't much going on behind the scenes anyway, so assigning to $du is just as valid. Thanks. Andrew ___ 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
Re: [SR-Users] Pseudo-variables for Status-Line
On 12/29/2011 03:36 PM, Robert R wrote: Thank a lot. $T_reply_code works. I tried all variables in pv doc ($rc, $err.rcode, $rs ... ) and none of them works, Actually the $rs pseudo-variable should also work as described here: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#sip_reply_s_status_status-code_response-code_reply-code Good that it helps. ___ 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
Re: [SR-Users] Pseudo-variables for Status-Line
On 12/29/2011 03:21 PM, Robert R wrote: What is the Pseudo-variables for Status-Line filed of SIP response messages (2xx, 3xx,4xx,5xx,6xx)? i.e., is there a Pseudo-variables to display the SIP response code? You can test it with tm function t_check_status(...) or get it in a config variable $T_reply_code: http://www.kamailio.org/wiki/cookbooks/3.2.x/pseudovariables#t_reply_code ___ 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
Re: [SR-Users] usrloc, timer process and cache cleanup
On 12/23/2011 08:18 PM, Stefan Sayer wrote: shouldn't the db layer and driver be smart enough to do insert ... on duplicate key update at least where it's supported? my fear is that such first insert then update policy will affect the performance. can create noise in the log on some db backends too.. ___ 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
Re: [SR-Users] TLS--UDP not working with record_route_preset
Gnani, since version 3.2.0 kamailio accepts several arguments to record_route_preset() function: http://www.kamailio.org/docs/modules/3.2.x/modules_k/rr.html#id2542169 In 3.1.0 you can call insert_hf directly, but IMO it's worth upgrading. Andrew On 12/08/2011 08:50 PM, Gnaneshwar Gatla wrote: Hello All, I have setup the Kamailio 3.1.0 behind a NAT/firewall. I’m using record_route_preset() function to keep the Public IP(advertised_address) in the Route header. I have to setup a where TLS-UDP and UDP-TLS to work. The sip clients do TLS and we have third party sip clients that only do UDP. The problem, Kamailio does not do double record-route when I provide record_route_preset(). This is leading to a problem where the call is either hung on TLS or UDP. I’m unsure how to go around this problem. I had posted earlier about this problem, is there a way I could use the advertised_address in the record route headers in this case? Regards Gnani ___ 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
Re: [SR-Users] transport=TLS
Hi Bruno, What I have done is an explicit check if (uri_param(transport,tls) || uri_param(transport,TLS)) to call force_send_socket with either udp or tls port. It would be cool for kamailio to select the proper socket automatically, I think there was a discussion on that previously but I can't find it right now. Hopefully someone else has a hint :) On 11/29/2011 02:50 PM, Bruno Bresciani wrote: Hi All, kamailio 3.1.2 recognize value transport=TLS on contact header or only transport=tls? kamailio forwards (t_relay function) the message with UDP protocol when value transport=TLS is on contact header. Cheers ___ 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
Re: [SR-Users] transport=TLS
Bruno, the address from contact header is put into R-URI on outgoing request to that user. This is where I catch that parameter. I think we should debug why kamailio sends the request using UDP, it is not clear, as Daniel pointed out it should work automatically. I think I had to do these manipulations because in my case the outbound proxy address is set On 11/29/2011 05:38 PM, Bruno Bresciani wrote: In my case the transport=TLS is present in contact header, has the same treatment of R-URI? Cheers ___ 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
Re: [SR-Users] How to limit number of calls in Kamailio proxy
Austin, the block beginning with dlg_manage() should be placed between route(AUTH) and route(PSTN). It is commented out and put outside the main routing block so I'm not sure if that was the case.. If the problem persists please get the dialogs list before calling, while the callee's phone is ringing after the call is established and after the call is cleared. If I am not mistaken the command is: kamctl fifo profile_list_dlgs caller On 11/05/2011 03:02 AM, Austin Einter wrote: Dear Andrew Here I am attaching the config file I am using currently. Please suggest me if something is wrong there. Also how do I limit the maximum number of calls. Thanks, Austin ___ 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
Re: [SR-Users] How to limit number of calls in Kamailio proxy
Austin, Actually you could share your config to the mailing list, I will tell you if there is something blatantly wrong ;-) Regards, Andrew On 11/03/2011 06:15 AM, Jason Penton wrote: Hi Austin, Have a look at the TM module docs. You will find the appropriate commands there. HTH On Nov 3, 2011 3:01 AM, Austin Einter austin.ein...@gmail.com mailto:austin.ein...@gmail.com wrote: Dear Andrew, Henning Thanks for sharing very useful information. I am bit new to kamailio, probably askingvery easy questions, please bear with me. I am not sure if I am forwording the INVITE statelessly or not. How do I check if INVITE is forwarded statelessly. Andrew you have mentioned 'So make sure that a transaction exists or create it explicitly using the tm module. Not sure, how do I check if transaction exists or not. And how can I create it. Please give me some pointers or a sample config file. Regards, Austin ___ 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
Re: [SR-Users] kamailio sip-capture error on compile
Jeff, On 10/03/2011 08:45 PM, Jeff Anderson wrote: I am trying to get kamailio sip-capture up and running for use with homer. I can get the service to start but i receive the following error. [root@homer kamailio]# /usr/local/sbin/kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(10208) WARNING: core [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 [...] Does anyone have any ideas on how to resolve this error? This message is from kamailio core, not sip-capture. I don't see anything abnormal in the log. Could you please share what are you doing and what makes you think sip-capture doesn't work? Regards, Andrew ___ 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
Re: [SR-Users] Account 403 to RADIUS
On 09/26/2011 02:16 PM, Mino Haluz wrote: Ok, the problem is the original feeradius package (squeeze) does not support Update radius messages (it gives Unsupported Acct-Status-Type = 15). It's not related to kamailio. .. however anybody knows how to patch it? The patches are here: http://download.dns-hosting.info/CDRTool/contrib/freeradius-brandinger/ but you can just grab freeradiusd-xs from the AG repository as described here: http://cdrtool.ag-projects.com/wiki/Install HTH. ___ 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
Re: [SR-Users] Nat problems
On 09/02/2011 08:58 PM, David Zambrano wrote: After doing some traces on the network I realized that the transcoder is trying to reach the router in front of the softphone and skipping the Kamailio loadbalancer. The call never reaches the softphone so the phonecall never completes. In UDP exactly the same thing happens, the call skips the Kamailio loadbalancer but the call completes perfectly fine. Well, you can not open the TCP connection to the client behind NAT. You need to reuse the existing connection created during registration, so the request should go via load balancer by record-route or path. I'm not sure why this works for UDP, probably your router is too permissive in this case :-) Regarding the loadbalancing and failover, I don't understand what is the problem. Could you specify which lb module do you use and preferable post the part of config responsible for failover. I believe you don't need to send request to both transcoders. To skip the transcoder that goes down you need to setup ping checks, if that is the only problem. ___ 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
Re: [SR-Users] Nat problems
On 09/02/2011 10:33 PM, David Zambrano wrote:
Hi andrew. Thanks for your help. What module or config should I use to
make sure the connection goes back through the loadbalancer?
That's simply the task for record-route like:
if (is_method(INVITE))
record_route();
But you also need the loose_route for routing new in-dialog requests.
For the failover I didnt specify anything ping related. Im using the
dispatcher module. Can I specify the ping config in that module or
should I use another module for that?
yes, it's in the documentation of dispatcher module:
http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html#id2806108
route{
ds_select_dst(2, 4);
t_relay();
}
well, with such config the chances are that the subsequent BYE may
arrive at the different server than the INVITE, so again you need the
loose_route section for this. You should get familiar with the default
config file to get a feeling of things.
Regards,
Andrew
___
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
Re: [SR-Users] issue with tm callbacks / premature cancel
Daniel, On 02.08.2011 11:35, Daniel-Constantin Mierla wrote: No, it does not (30 150). I've also tried settings max_inv_lifetime to even greater 300 seconds, no luck.. OK. Btw, do you set both parameters of t_set_fr(...)? I've tried adding a second parameter, also inserted t_reset_fr() before the call to t_set_fr(), still no luck. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] issue with tm callbacks / premature cancel
Hi Daniel, On 01.08.2011 20:12, Daniel-Constantin Mierla wrote: does it happen to exceed the max lifetime for transaction? http://kamailio.org/docs/modules/stable/modules/tm.html#max_inv_lifetime No, it does not (30 150). I've also tried settings max_inv_lifetime to even greater 300 seconds, no luck.. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] Set CLI.
Hello Henrik,
On 28.07.2011 19:08, Henrik Aagaard Sørensen wrote:
Is it possible to set CLI via the usr_preferences table in Kamailio?
Or any other way?
So that certain subscribers get there from-number overwritten with a
specific one.
Sure, you can script it in any way you like.
This is a working example from my system:
if (avp_db_load($au, $avp(s:allowed_cli))) {
if (!avp_check($avp(s:allowed_cli), eq/$fU/gi)) {
xlog(L_INFO, User $au is denied CLI=$fU\n);
sl_send_reply(403, Forbidden);
exit;
}
}
You see, I'm rejecting calls with a CLI that is not allowed.
To override CLI you may use something like this, probably this should be
put directly before call to t_relay():
# Replace from if needed
if(is_avp_set($avp(s:allowed_cli))) {
uac_replace_from(,$avp(s:allowed_cli));
xlog(L_INFO, $ci : replaced from to $avp(s:allowed_cli)\n);
}
--
Sincerely,
Andrew Pogrebennyk
___
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
Re: [SR-Users] kamailio + rtpproxy behind nat possible?
What are you using - rtpproxy, mediaproxy etc? There is an experimental patch for advertised_address support in rtpproxy on the internet, but I've never had a chance to try it out. You may want to try spce-v2.2, which comes with mediaproxy-ng supporting the advertised_address out of the box: http://www.sipwise.com/news/announcements/spce-v2_2-release/ On 29.06.2011 16:56, MingHon wrote: will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] Kamailio and DTMF
Phillip, This question is irrelevant to Kamailio. The DTMF mode and codecs are negotiated between the two endpoints only, so Kamailio supports any codec. On 28.06.2011 11:06, Phillman25 Kyriacou wrote: Hi I was wondering what DTMF modes Kamailio supports? If i wanted to force a DTMF mode on Kamailio how could i go about doing it? thanking you in advance Phillip -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] Kamailio and DTMF
On 28.06.2011 11:18, Raúl Alexis Betancor Santana wrote: This is not fully true on the DTMF side, if you use SIP-INFO as DMTF transport, DTMF's will go throught kamailio (if configured to do so). Right - but then this is achieved through the configuration script. Andrew ___ 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
Re: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service when load the ldap.so module
On 21.06.2011 17:08, laura testi wrote: Yes, it can be read by all user like other configuration file: -rw-r--r-- 1 root root 566 Jun 20 18:56 ldap.cfg Just in case, have you set perhaps any of the environment variables manually for working with ldap? 'service' command is a wrapper that calls env -i (start with an empty environment), maybe this is the reason it works differently for the root and kamailio users. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] incorrect port 0 in reply from rtp proxy
Mokhtar, could you please make sure that you are calling route(RTPPROXY) in reply route as well, as Alex suggested? On 03.06.2011 16:33, Mokhtar Bengana wrote: This is how I configured rtpproxy. Not sure why rtpproxy is not engaged both ways. Thanks for your help. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] Questino about dialplan module
On 03.06.2011 22:58, Gary Chen wrote:
reload before you can see the change. Why the ' kamctl dialplan show'
display the data directly from mysql database? Does that mean that
dialplan data is not stored in the memory?
Gary, a quick look at kamctl script proves that it displays the data
directly from the mysql database, not from memory:
# #
### DIALPLAN management
#
dialplan() {
require_dbengine
require_ctlengine
case $1 in
show)
shift
if [ $# -gt 0 ] ; then
mecho dialplan $1 tables
QUERY=select * FROM $DIALPLAN_TABLE WHERE
$DIALPLAN_DPID_COLUMN=$1 ORDER BY $DIALPLAN_PR_COLUMN ;
else
mecho dialplan tables
QUERY=select * FROM $DIALPLAN_TABLE ORDER BY
$DIALPLAN_DPID_COLUMN, $DIALPLAN_PR_COLUMN;
fi
$DBROCMD $QUERY
;;
--
Sincerely,
Andrew Pogrebennyk
___
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
Re: [SR-Users] Kamailio + Asterisk 1.6.2 Realtime - REGFWD : 401 not authorized
Skyler wrote: After 3 hours stuck on this I have to ask the group. I am setting up Kam 3.1.3 +Ast 1.6.2.18 + realtime following Daniel’s guide on the Asipto site. The problem I see is 401 not authorized when uac tries to register. Hey, kamailio is addictive - there are quite some of us doing it on Sundays ;-) I've just checked the manual, it says kamailio, not asterisk should do authentication of REGISTERs. Could you check you have created sipusers table and configured asterisk as per manual: sipusers is the standard table required by Asterisk to store SIP user profile, with one extra column sippasswd where will be stored the password for SIP authentication. By default, Asterisk uses the column secret for SIP user password, but if that is filled in, Asterisk will ask for authentication again, resulting in double-authentication which we want to avoid. ? -- Sincerely, Andrew Pogrebennyk ___ 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
[SR-Users] how to add RR with advertised_address in front of normal RR?
I've been playing with sip load-balancer in EC2 but so far haven't been able to figure out how to insert Record-Route header with advertised_address. AFAIK kamailio rr module adds 2 RR headers (one with the inbound address and one with the outbound address) when your transaction comes with one interface/port and leaves via another. In EC2 the receive socket is the same as send socket and I've had no luck trying to get kamailio insert advertised address as inbound RR and then the outbound RR, by calling record_route_preset() several times in a row too. This is a problem if lb is in front of another proxy. As I see this question has been raised on the mailing lists as far as 6 years ago - I would be surprised if no one has come up with a solution yet. So how to do record-routing in a draft-ietf-sip-record-route-fix AKA rfc5658 style on the machine with one physical interface? -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] SIP Trunk Usage Monitoring
On 09.05.2011 22:44, JR Richardson wrote:
Would there be any usage examples of dialog module? I'm not sure if I
really need profiles and [values] and when to set and unset. A push
in the right direction would be helpful.
Hi,
here is an example I've used for incoming and outgoing calls tracking,
just tune it for your purposes:
loadmodule dialog.so
modparam(dialog, dlg_flag, 4)
modparam(dialog, db_mode, 1)
modparam(dialog, profiles_with_value, carrierin;carrierout)
...
create_dialog();
if(is_from_gw())
{
# Monitor number of incoming calls
get_profile_size(carrierin,$si,$avp(s:cnt));
xlog(L_INFO, - currently, the carrier $si has
$avp(s:cnt) active outgoing calls\n);
set_dlg_profile(carrierin,$si);
} else {
# Monitor number of outgoing calls
get_profile_size(carrierout,$rd,$avp(s:cnt));
xlog(L_INFO, - currently, the carrier $rd has
$avp(s:cnt) active outgoing calls\n);
set_dlg_profile(carrierout,$rd);
}
--
Sincerely,
Andrew Pogrebennyk
___
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
Re: [SR-Users] BLF using kamailio
Lucas, It's not possible to answer your question without at least a corresponding SUBSCRIBE message from Aastra. On 20.04.2011 17:17, Lucas Alvarez wrote: Hi, I have implemented kamailio with asterisk, I want to enable BLF using kamailio instead of asterisk. I would appreciate any kind of help, below I'm pasting my configuration. *I'm testing with an Aastra phone, when phone sends the subscription kamailio answers with:* U 192.168.15.22:5060 http://192.168.15.22:5060 - 192.168.15.108:5060 http://192.168.15.108:5060 SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.15.108;branch=z9hG4bKa54a155ad67706dca. From: 1104 sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060;tag=71cf5c9949. To: sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060;tag=99b712eabe88b2bf7a20409f3dc7ebf1-f71b. Call-ID: f22823b8b4d58a8e. CSeq: 3430 SUBSCRIBE. Allow-Events: presence.winfo, presence, dialog. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] BLF using kamailio
Lucas, It seems the Aastra is not configured for BLF, the SUBSCRIBE below is for message-summary, not dialog info. Kamailio is sending the list of supported events in Allow-Events: presence.winfo, presence, dialog On 20.04.2011 17:45, Lucas Alvarez wrote: Thank you Andrew for your quick answer. *This is the subscribe message:* U 192.168.15.108:5060 http://192.168.15.108:5060 - 192.168.15.22:5060 http://192.168.15.22:5060 SUBSCRIBE sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.15.108;branch=z9hG4bKa54a155ad67706dca. Proxy-Authorization: Digest username=1104,realm=192.168.15.22,nonce=Ta7q1k2u6ar6DmLu8tDlXtvfjjhWEBct,uri=sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060,response=de6b80891ba1923830cefc552b47. Max-Forwards: 70. From: 1104 sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060;tag=71cf5c9949. To: sip:1104@192.168.15.22:5060 http://sip:1104@192.168.15.22:5060. Call-ID: f22823b8b4d58a8e. CSeq: 3430 SUBSCRIBE. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO. Allow-Events: talk, hold, conference, LocalModeStatus. Contact: 1104 sip:1104@192.168.15.108:5060;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D10A1A6. Event: message-summary. Expires: 86400. Supported: path. User-Agent: Aastra 57i/3.2.1.43 http://3.2.1.43. Content-Length: 0. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] topoh, angle brackets and Contact URI params interpretation
Daniel, The problem is fixed and the topoh module has been running fine for a few hours so far. Thank you. On 04.02.2011 20:32, Daniel-Constantin Mierla wrote: Hello, I added the code for enclosing contact uri between angle brackets. I committed on git master branch, but have no way to test it for now due to traveling. If you can test it and report back would be great. When all is working fine, I will backport to 3.x branches. Thanks, Daniel -- Sincerely, Andrew Pogrebennyk ___ 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
[SR-Users] topoh, angle brackets and Contact URI params interpretation
I'm having an issue with topoh module in Kamailio 3.1.0. When Contact header is formed like this: Contact: 0991 sip:192.168.0.107;line=sr-N6IAzBysz.tyz.D4M.VLOBMfOBFuWxvfMxV4 The other party responds properly. But when there is no angle brackets in Contact: Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* Really, many parsers implement prefer shift to reduce principle, which means if something can be interpreted in more enclosed expression, it will be interpreted this way and no as part of less enclosed expression so ;line is interpreted as header parameter but not URI parameter. RFC 3261 section 20 suggests that any URI parameters be contained within angle brackets: === The Contact, From, and To header fields contain a URI. If the URI contains a comma, question mark or semicolon, the URI MUST be enclosed in angle brackets ( and ). Any URI parameters are contained within these brackets. If the URI is not enclosed in angle brackets, any semicolon-delimited parameters are header-parameters, not URI parameters. === I think the topoh module should force the angle brackets. BTW it seems that parameter needs to be urlencoded, see rule 'other-param' in RFC 3261 section 25.1: other-param = pname [ = pvalue ] pname = 1*paramchar pvalue= 1*paramchar paramchar = param-unreserved / unreserved / escaped param-unreserved = [ / ] / / / : / / + / $ No .-* characters are allowed in the paramchar. But at least that's not causing me any problems. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] [sr-dev] topoh, angle brackets and Contact URI params interpretation
On 03.02.2011 10:26, Andrew Pogrebennyk wrote: I think the topoh module should force the angle brackets. BTW it seems that parameter needs to be urlencoded, see rule 'other-param' in RFC 3261 section 25.1: From what I understand the valid form is: Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* or Contact: sip:192.168.0.107;line=sr-N6IAzB3AWxyfz.stM.quOBFZMJZfWxj7W.y-MljAWBy* so it should be enclosed by angle brackets or double quote, otherwise most implementations would treat ;line as header parameter and the parsing would fail since @ is not allowed as header parameter value if it's not enclosed by double quotes. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] [sr-dev] topoh, angle brackets and Contact URI params interpretation
On 03.02.2011 18:12, Daniel-Constantin Mierla wrote: I will check the sources and fix if the contact address is not between . However, I do not undeerstand where you got the @, is none there or am I missing something? Daniel, Sorry, I meant the * sign. Thanks for looking into it. -- Sincerely, Andrew Pogrebennyk ___ 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
Re: [SR-Users] SIP Softphone with RPID and PAI
On 24.11.2010 22:02, Uriel Rozenbaum wrote: :) I know is not the answer I expected, but I'll do something like that, maybe use sipp or sipsak. Uriel, sipp or sipsak will do just fine, I've also used SIPp for this particular purpose: http://sipper.agnity.com/ - useful for more complex scenarios too (and I have no relation to the authors whatsoever:)). -- Sincerely, Andrew Pogrebennyk ___ 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
[SR-Users] correct way to define flags in Kamailio 3.0.3
What is the correct way to write flags in Kamailio 3.0.3? I've tried both enum-like way: flags a, b; and the macro way: #!define a 1, but neither has worked. I'm getting a syntax error trying to used defined flag, e.g.: modparam(usrloc, nat_bflag, FLB_NATB) In 3.1 though the macro works fine. -- Sincerely, Andrew Pogrebennyk ___ 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
[SR-Users] SER-like domain attributes in Kamailio?
I'm trying to migrate SER 2.0 to Kamailio and right now the major hiccup is migrating lookup_domain(). Is there any replacement for lookup_domain() that would enable you to use the domain attributes in Kamailio? I have come across the old thread domain attributes about adpoting the SER domain module: http://www.mail-archive.com/sr-...@lists.sip-router.org/msg00804.html But I'm not sure what was the outcome. Same for load_attrs() and user_attrs column - it looks like there's no direct replacement for that and I have to migrate it to usr_preferences. -- Sincerely, Andrew Pogrebennyk ___ 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
