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37560 discussions

t_save_lumps() and failure_route
by Julien Chavanton 14 Jun '18

14 Jun '18

Hi, I have a scenario where I am using. t_save_lumps(), lookup() and then I can choose to drop some branches in branch routes. However, it is possible that I endup without any branch left, in this case t_relay() is returning and no failure route is called/created. At this point I need to recover the original message I believe it is not possible to restore/reset the message without passing by failure route ? Else, I could trigger a failure route using t_relay(), is there any "handy/clean" way to trigger failure_route() ? Regards

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Memory leak in tm with push notifications
by Ivaylo Markov 14 Jun '18

14 Jun '18

Hello, I am trying to set up Kamailio as a push notifications proxy, closely following the example in the "Kamailio in a Mobile World" presentation (https://www.slideshare.net/FedericoCabiddu/kamailioinamobileworld-51617342). I am running Debian 9 and Kamailio 5.1.3 from the official Debian repositories. I believe the main modules involved in the issue below are tm, tmx, and tsilo. Every call passing through the proxy leads to a small memory leak in the tm module - there is a large amount of "delayed free" memory cells from tm's internal hash table. At some point the shared memory runs out and Kamailio restarts. Using the "kamcmd corex.shm_summary" command I was able to see that the top users of shared memory are "tm: h_table.c: build_cell" and "core: core/sip_msg_clone.c: sip_msg_shm_clone" with the same allocation count. I experimented with removing different parts of the configuration and noticed that commenting out the "t_continue(...)" call in the "PUSHJOIN" route (see slide #22) prevents the leak from happening. Maybe something in that function is incrementing the reference counter to the hash table cell, but it is not decrementing the counter when done? I tried looking around the source code of the tm and tmx modules, but saw nothing suspicious. I also tried using gdb with a breakpoint in t_continue_helper (tm/t_suspend.c:166) hoping to see what else is accessing the htable cell, but was unable to find anything of use. Has someone encountered anything like this? Can you provide more directions on debuggin this? I can provide some bits of configuration, but an entire test setup would be rather difficult, unfortunately. Thank you for your time, Ivo

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integrate huawei HSS with kamailio
by eyas barhouk 14 Jun '18

14 Jun '18

hello dears i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS , so is there any one has an experiment with this or tested it before ??? if yes, could you please guide me how to do that ? or where to start ? thanks in advance

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Forbidden - Private identity not found (Authorization: username)
by eyas barhouk 14 Jun '18

14 Jun '18

Hello dears when i was trying to build an IMS platform by using kamailio with sip digest authentication scheme and ims_auth module, i couldn't register and the result was as following : 1-the first register message passed to the s-cscf then S-cscf replaying with the message “401 unauthorized - challenging the ue” 2- the second register pass to the S-cscf with the authorization header then the S-CSCF replying with "Forbidden - Private identity not found (Authorization: username)" So based on that could any one tell me what is the reason of this error and how to fix it ?? Thanks in advance

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Using a var as timeout value in dlg_set_timeout_by_profile function
by jenus＠cyberchaos.nl 13 Jun '18

13 Jun '18

I'm trying to use the dlg_set_timeout_by_profile in the xhhtp route to update the dialog timeout for active calls. But if i use a var as the timeout value the call disconnects immediately. This works fine (call disconnects after 80 sec): dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "80"); This fails (call disconnects immediately: $var(DialogTimeout)= 80; dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "$var(DialogTimeout)"); Are vars supported in the dlg_set_timeout_by_profile function? And is it possible to use it in any route as described in the docs? I'm running kamailio 5.1.3. Thanks, Jan

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Registration of UACs i n a partially encrypted scenario
by mip FKF 13 Jun '18

13 Jun '18

Hello, we are pretty new to SIP and kamailio, we do have some questions regarding the following scenario: We have a number of UACs in a small network which are required to communicate without encryption because the are not able to consume certificates. We want to use kamailio (as a proxy?) to establish an encrypted connection to a backend UAS. 1. Is it possible to directly register the UACs with the UAS eventhough communication between kamailio and the UAS is encrypted ? 2. How do we need to configure kamailio in order to make this scenario work ? Thank you for your suppport

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Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX
by Floimair Florian 13 Jun '18

13 Jun '18

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint (chan_pjsip). With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of "Wilkins, Steve" <swwilkins(a)mitre.org> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 12. June 2018 at 16:27 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can’t be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

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Asterisk inviting Kamailio
by Wilkins, Steve 13 Jun '18

13 Jun '18

Hello All, I have noticed that sometimes when a call is made from one endpoint to another through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after the call has been established. Sometimes this does not happen. When it does happen, calls drop. Why would an INVITE be sent back to Kamailio? Also, Pan, I am working on the response you requested yesterday. Thanks you All, -Steve

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 12 Jun '18

12 Jun '18

Got it working. Thank you everyone. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 3:06 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< Alex, Pan, Daniel,... Could this group => group:BUNDLE audio video in Message Body have anything to do with my Kamailio Video issue. Thank you!! -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 12:14 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Hi Alex, No, I'm not using rtpengine. It must definitely be some sort of Codec issue though, since I can seem to move the problem around and actually finally get video on the softphone. It was a little strange that I lost Audio on the WebRTC client though, since I only disabled VP8. Since Kamailio is only relaying I am so confused why introducing Kamailio is messing up Video. Thank you, -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov Sent: Sunday, June 10, 2018 11:46 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Unless you're using rtpengine, you're just moving the problem around. Kamailio does nothing with SDP unless by way of rtpengine or told in some other way, such as sdpops. On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" <swwilkins(a)mitre.org> wrote: >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I >now get Video on the softphone however, I lost two-way Audio. Kamailio >seems to be doing something with the codecs but still can't put my >finger on it. The WebRTC Client, who is the caller, needs VP8 for >Video, and apparently Audio. > >I'm not sure if I getting closer or just moving the problem around. > >Thanks All! > >From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf >Of Wilkins, Steve >Sent: Saturday, June 9, 2018 12:47 PM >To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when >using Kamailio...works without Kamailio > >>>>Email originates from a non-MITRE system. Use caution.<<< >I am desperately trying to resolve a big issue I have when using >Kamailio 5.2.0 and Asterisk 15.3. > >As soon as I go through Kamailio, I get no Video on either side of the >call; I do get two-way audio and the call stays connected. If I simply >go right through Asterisk, I also get two-way Video. I am having a >tough time determining why Kamailio is messing with the Video portion >of the call. >I have included a full pcap file. As a side note when comparing pcap >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from >Asterisk and Softphone are exactly the same. > >Thank you, >-Steve -- Alex -- Sent via mobile, please forgive typos and brevity. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Question on Contact Header on Kamailio 200 OK to PBX
by Wilkins, Steve 12 Jun '18

12 Jun '18

Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can't be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

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