sr-users February 2017

sr-users@lists.kamailio.org

74 participants
99 discussions

Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

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Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

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want to see active users SIP users
by Mandar Nagarkar 24 Feb '17

24 Feb '17

Hello Everyone, I want to see active users SIP users for this purpose I checked the location table of kamailio database it does have active users entries however as per my observation there is 45 secs to 1 min delay before the entry actually added to this table i.e. suppose if I am registering the user through pjspi,jssip or any client I cannot see entry for that user in location table. I have to wait up-to 1 min to see the user in location table. Is there any way we can Identify if user is online without delayed entry? Regards, Mandar nagarkar The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

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Fetching proxy-autheticate header from 407 Response
by vivek＠advaitamtech.com 23 Feb '17

23 Feb '17

Hi All, Am not using the authentication in the kamailio. So I will be receiving the "407 Proxy Authentication Required" response from the third party server for the first INVITE. I want to fetch the Proxy-Authenticate header present in the 407 response. I browsed and tried using the pseudo-variables "$adu/$aa/$ar". It dint work. Please let me know if there is a way to fetch this header. Thanks, Vivek.

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How to release call without having to use fromTag and toTag, and resume a suspended call
by Cassandra Yurekli 23 Feb '17

23 Feb '17

To whom it may concern, There are several points I don’t quite understand regarding a couple of kamailio modules. I have a main application which controls calls by sending command messages via a socket (for example reject this sip invites). I communicate with the socket client via evapi module and using evapi_asynch_relay() in order to be sure that communication are suspended and kamailio waiting an order for doing something with them (Accepte, reject, release, ...). I want to release a call, using a command “release this call”. I’m actually doing this with dialog module (using get_dialog(callid,fromTag,toTag) and dlg_bye(‘all’)) but is there anything else (like another module) which could release a call without necessarily having to use fromTag and toTag to get the context ? I didn’t find it on modules documentations. Also after an INVITE I’m waiting for the main app to accept or reject the call. For accepting and continuing the transaction I found “t_continue(tindex,tlabel,route)” on tmx module, but at this moment $T(id_index) and $T(id_label) are null, is it normal or I did something wrong ? How can I simply resume the suspended call ? Thanks; Best regards,

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Need help with tcpops
by Abdoul Osséni 22 Feb '17

22 Feb '17

Hi, My call flow is: Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN I want to use tcpops module between the softphone and kamailio. http://www.kamailio.org/docs/modules/4.4.x/modules/tcpops - enable tcpops : no problem - disable tcpops : two cases - cancel or bye from softphone: no problem. - cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment. Regards Abdoul.

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Automated integration tests for individual modules
by A Lai 22 Feb '17

22 Feb '17

Hi, I just started working with the Kamailio code base. I am wondering if there are some common strategies in having automated tests on proper functioning of modules when loaded into cfg.

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problem with avpops and mongodb modules
by hsdsdssdfsdf dsasd 22 Feb '17

22 Feb '17

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How to handle CANCEL if corresponding INVITE transaction is missing
by Venkatesh Macha 22 Feb '17

22 Feb '17

Hi all, I am using Kamailio 4.4.5, It is working fine. But in one case. If i make a video call between the two users and cancels the call then Call is not disconnecting on the callee side. Please look the following scenario. 1000 made one video call to 2000 2000 is ringing. Now 1000 disconnected the call by sending CANCEL. But 2000 is not disconnecting. This is happening because Kamailio is unable to find the corresponding INVITE transaction for the CANCEL so it is simply dropping the requests. Why Kamailio unable to find the transaction. And is there any possibility to handle this case and send the CANCEL to callee side. I tried with forwad() and someother things but it is just generating the loops. I am also attaching the ngrep trace. Thank you in Advance. CANCEL_not_working_nabble.CANCEL_not_working_nabble <http://sip-router.1086192.n5.nabble.com/file/n156243/CANCEL_not_working_nab…> -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-handle-CANCEL-if-correspondi… Sent from the Users mailing list archive at Nabble.com.

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sr-users February 2017