- sr-users - kamailio.org

Difference between HTTP clients (utils.so vs http_client.so)
by Joel Serrano | VOZELIA 12 Jan '17

12 Jan '17

Hi all, Currently I see there are 2 http clients: utils.so --> http_query() http_client.so --> http_client_query() Any advice on which to use? Any pros/cons of using one over the other? Thanks, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

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RPC exact command to change voicemail IP or Port
by Annus Fictus 12 Jan '17

12 Jan '17

Hello list, I can't find the exact RPC command to change IP or port of voicemail server (module CFG_RPC): the default Kamailio configuration is: */#!ifdef WITH_VOICEMAIL/**/ /**/# VoiceMail Routing on offline, busy or no answer/**/ /**/#/**/ /**/# - by default Voicemail server IP is empty to avoid misrouting/**/ /**/voicemail.srv_ip = "" desc "VoiceMail IP Address"/**/ /**/voicemail.srv_port = "5060" desc "VoiceMail Port"/**/ /**/#!endif/* Thank you Regards

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how to immediately close a tls connection
by Andy 12 Jan '17

12 Jan '17

Hi, trying to close a tls connection immediately, have ref to the connection (tcpconn) pointer please suggest, thanks! -- View this message in context: http://sip-router.1086192.n5.nabble.com/how-to-immediately-close-a-tls-conn… Sent from the Users mailing list archive at Nabble.com.

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Kamailio and default paramenters to a specific gatewy
by Laura 11 Jan '17

11 Jan '17

Dear all, I'm using kamailio 4.3.6 and LCR modules.. i was wondering if it was possibile to add some special paramenter to all call sent to a specific gateway. I was try to use the " params" field on lcr_gw table but it's not working like i was think.. The issue is simple.. i want to add to field "Privacy: id" to any call i send over a specific gateway.. What is the best and easy way to do that ? Any idea is appreciate. Regards Laura

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Diameter AVP handling from kamaiio.cfg
by Zoltán Miricz 10 Jan '17

10 Jan '17

Dear All, I'm wondering is it possible to catch Diameter Messages and their AVP values from kamailio.cfg just like for SIP messages? (Using pseudo variables) I looked into this function from the attached S-CSCF example. route[REG_MAR_REPLY] { #this is async so to know status we have to check the reply avp xlog("L_DBG","maa_return code is *$avp(s:maa_return_code)*\n"); Sadly it gives back only 1, -1, -2. (which is based on cxdx_mar.c the rc) What I mainly looking for is the EXPERIMENTAL_RESULT (experimental_rc) and some different AVP values besides. Thank you very much in advance! *Cheers,* *Zoltan*

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Kamailio on Cluecon Weekly – Jan 11, 2017
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, for those interested in participating to a (video) chat tomorrow, Wed, Jan 11, I will be joining the ClueCon Weekly at 12:00CT (USA) to discuss about latest Kamailio, what's new and the plans for the near future. Integration with FreeSwitch is going to be among the topics. You can join the video conference call with a webrtc capable browser. You can also join only for audio using a SIP phone or dialing from PSTN or mobile networks. More specific details are available at: - https://www.kamailio.org/w/2017/01/kamailio-on-cluecon-weekly-jan-11-2017/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Daniel-Constantin Mierla 10 Jan '17

10 Jan '17

Hello, are you using topoh module? A pcap with all sip messages part of the dialog, from the initial INVITE to the last BYE transaction would be important to have in order to investigate how the headers are changed. Cheers, Daniel On 28/12/2016 13:11, Nihar Ranjan Deb wrote: > > Hi All, > > > > I am a new user of kamailio. We have setup a kamilio server with RTP > proxy. Most of the features are working fine except BYE. > > > > > > Say User A and B are registered in kamailio server. A (caller) > connected to B(called), call established. > > > > 1. If Caller (A) disconnects the call, everything goes fine. B gets > disconnected and call releases. > > > > 2. But if Called party (B) disconnects the call, A never gets > disconnected. I checked trace as well, in this case caller (A) gets a > BYE with B's callid. So A rejects call with "481 call leg transaction > doesn't exists". > > Basically A party and B party has separate dialog established with > Kamilio. So they have two separate callids . All messages are > exchanged correctly. Only when B party disconnects call, kamailio > sends BYE to A party with B party callid. > > > > Please help me to resolve this issue. Do let me know if any more > information required. > > > > > > ------ INVITE-------------------------- > > > > INVITE sip:966268@*** SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport > > Max-Forwards: 70 > > Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> > > To: <sip:966268@***> > > From: "334031"<sip:334031@****>;tag=4406552c > > Call-ID: *82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc* > > CSeq: 2 INVITE > > Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, > OPTIONS, MESSAGE > > Content-Type: application/sdp > > Proxy-Authorization: Digest > username="334031",realm="*****",nonce="****",uri="sip:966268@***",response="***",algorithm=MD5 > > Supported: replaces > > User-Agent: X-Lite release 4.9.6 stamp 82158 > > Content-Length: 334 > > > > v=0 > > o=- 13127393523589215 1 IN IP4 192.168.1.36 > > s=X-Lite release 4.9.6 stamp 82158 > > c=IN IP4 192.168.1.36 > > t=0 0 > > m=audio 57116 RTP/AVP 9 8 120 0 84 101 > > a=rtpmap:120 opus/48000/2 > > a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 > > a=rtpmap:84 speex/16000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > > > > > > -------------------------BYE-------------------------------- > > > > > > BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > From: <sip:966268@***>;tag=1665200125 > > To: "334031" <sip:334031@***>;tag=4406552c > > Call-ID: > *!!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV*** > > CSeq: 3 BYE > > Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> > > Max-Forwards: 69 > > User-Agent: Yealink SIP-T20P 9.72.0.80 > > Content-Length: 0 > > > > > > -------------------------- 481--------------------- > > > > SIP/2.0 481 Call/Transaction Does Not Exist > > Via: SIP/2.0/UDP > ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.164.228.12 > > Via: SIP/2.0/UDP > 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* > > To: "334031" <sip:334031@***>;tag=4406552c > > From: <sip:966268@***>;tag=1665200125 > > Call-ID: > !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRweMH4N6RemHYMI9Y0tV** > > CSeq: 3 BYE > > Accept-Language: en > > Content-Length: 0 > > > > > > > > > > Regards, > > Nihar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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[Error] Install Kamailio 4.4 in Mac 10.12
by Luis De Marchi 10 Jan '17

10 Jan '17

Hi, Sorry, my english, I'm using the translator. I'm having trouble installing Kamailio for Mac 10.12 Terminal: Luis:kamailio luisdemarchi$ sudo make all CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:41:10: fatal error: 'mysql.h' file not found #include <mysql.h> ^ 1 error generated. make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Luís De Marchi Diretor Administrativo +55 (41) 99815-6800 +55 (41) 3027-7669 www.snowmanlabs.com <http://www.snowmanlabs.com/>

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t_precheck_trans
by Annus Fictus 10 Jan '17

10 Jan '17

Hi, where can I find documentation for this function present on the default script? In TM module not exist (Kamailio 4.4.4) Thank you Regards

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use case
by Thufir Hawat 09 Jan '17

09 Jan '17

I'm ok with Linux and am dipping my toe into Asterisk by running an AWS EC2 instance. What is the most frequent usage for kamailio? Would it mainly be for SIP to SIP? So that the URI would be for a specific domain? thanks, Thufir

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using kamailio to forward ipv6 to ipv4
by anfecora 09 Jan '17

09 Jan '17

Hi guys hope you have a great new year. I would appreciate if anyone can point me in to the right direction . I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. is that possible? thanks.

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why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

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Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 09 Jan '17

09 Jan '17

Anyone any help on this. Regards, Nihar From: Nihar Ranjan Deb [mailto:nihar.ranjan@microtalkgroup.com] Sent: Wednesday, December 28, 2016 5:42 PM To: 'sr-users(a)lists.sip-router.org' Subject: BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists". Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

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VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

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

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presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

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Re: [SR-Users] using kamailio to forward ipv6 to ipv4
by anfecora 05 Jan '17

05 Jan '17

Yes i am planning on proxing media. The case is that client will only talk ipv6 but the registrar only have ipv4 connectivity. Therefore need to create a proxy to translate this dialogs. Thi thnking on using kamailio to receive the registration modify the pckets and forward registrations in ipv4. I was thinking in using the asterisk kamailio example but instead of asterisk behind will be an sbc. Therefore kamailio will listen in ipv6 and translate over to ipv4 including media of course via rtpproxy. On Jan 5, 2017 9:43 AM, "Frank Carmickle" <frank(a)carmickle.com> wrote: > On Jan 5, 2017, at 12:24 PM, anfecora <anfecora(a)gmail.com> wrote: > > Hi guys hope you have a great new year. > I would appreciate if anyone can point me in to the right direction . > > I need to build a proxy to translate from ipv6 to ipv4, but kamailio should not process the registers or invites, only translate from ipv6 to ipv4 and forward all registrations over. A bit more information about the situation would be helpful. What exactly are you trying to do? > is that possible? 6 to 4 translation is possible, but until we understand the situation I’m not sure we can be that helpful. Are these phones? If so are they dual stack? Is the underlying network dual stack? There’s a lot of work to do to get this right in a Kamailio config. You’ll need to run rtpengine to proxy the media if the endpoints are not dual stack. Who keeps the mapping of the v6 to v4 and v4 to v6 translations? How is that mapping determined? I’d recommend ds-lite if you are in a position to alter the network and the phones/systems allow for it. HTH —FC > thanks. > > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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CDR - acc_cdrs
by Annus Fictus 05 Jan '17

05 Jan '17

Hello, I'm trying to save Kamailio CDR on acc_cdrs table without success. My relevant configuration look like: ----- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 ----- #!define WITH_ACCDB ---- loadmodule "dialog.so" loadmodule "acc.so" modparam("dialog", "dlg_flag", 0) ---- # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_log_enable", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$td") #!endif Any hint is really appreciate. Regards

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unexpected char - status 6
by AL S 05 Jan '17

05 Jan '17

Hi, Kamailio is throwing error message from "parse_to.c" with status=6. The source code shows status 6 as URI_ENCLOSED. However, tcpdump of the call does not show any enclosed angle bracket is missing. (tcpdump pcap file is attached.) Kamailio log Jan 4 10:18:21 orl-ar-openser01 /usr/sbin/kamailio[2374]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK04001cd3>> . FROM and TO in INVITE received by Kamailio From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161>;tag=gK04001cd3 FROM and TO in INVITE forward by Kamailio for the same call From: <sip:7272697443@71.44.15.84>;tag=gK00d6c4e0 To: <sip:2057030997@72.31.12.161\000\000>;tag=gK04001cd3 PS: this error message is happening on all 4 kamailio servers (3.3.0) PS: call failure only happens in re INVITE (call stays active longer than 15 min) Your help would be appreciated. Thanks, AS

2 2

RTPProxy with VPN
by Rodrigo Moreira 05 Jan '17

05 Jan '17

Hello, I am having problems with RTPPROXY. I need it to work as a bridge. On one side will have a VPN tunnel called a tun0, on the other an eth0 interface. Is it possible for RTPPROXY to run correctly in this scenario? How can I test if RTPPROXY is running correctly? Is there any command? What would the configuration look like inside /etc/default/rtpproxy Please help me. Best regards -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Re: [SR-Users] Regarding usage of Redis Module in Kamailio core
by Vicente Hernando 05 Jan '17

05 Jan '17

Hello Hemanth, this minimal kamailio cfg file works for me to test ndb_redis module. #!KAMAILIO debug=4 log_stderror=yes children=4 fork=yes #disable_tcp=yes mpath="modules/" loadmodule "ndb_redis.so" # An already configured redis server is needed here modparam("ndb_redis", "server", "name=srv1;addr=127.0.0.1;port=6379;db=0") request_route { redis_cmd("srv1", "SET foo bar", "r"); exit; } Changing the parameters to suit your configuration, you can use it to debug your system. Regards, Vicente. On 01/05/2017 07:55 AM, hemanth(a)advaitamtech.com wrote: > > Hello Vicente, > > Thanks for your response, > > In response to your query, > > Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, > > I have also modified the kamailio.cfg file loadmodule and modparam list, > > I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. > > Thanks & Regards > Hemanth > > > Message: 8 > Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) > From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core > Message-ID: <1483521102.18025940(a)apps.rackspace.com> > Content-Type: text/plain;charset=UTF-8 > > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > > > ------------------------------ > > Message: 9 > Date: Wed, 4 Jan 2017 10:50:05 +0100 > From: Vicente Hernando <vhernando(a)systemonenoc.com> > To: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio > core > Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> > Content-Type: text/plain; charset=utf-8; format=flowed > > Hello Hemanth, > > just a quick response, I would check if you compiled ndb_redis module > correctly (also if you have hiredis library dependency). > > Also I would check if you loaded ndb_redis module in your configuration > file. > > > Regards, > Vicente. > > On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: >> Hello, >> >> I am using Kamailio as a SIP proxy, >> >> I want to store Registered user information in Redis DB, >> >> I am able to configure REDIS DB, But i am not able to run the redis command from >> >> Kamailio core using the function >> >> "redis_cmd("srvN","SET key 1", "r")" >> >> I am getting undefined reference error as below >> >> "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" >> >> can anyone please help me to fix this issue >> >> Thanks & Regards >> Hemanth >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ------------------------------ > > End of sr-users Digest, Vol 140, Issue 3 > **************************************** > > > >

1 0

Kamailio cancel branch only after receiving replies
by Aqs Younas 05 Jan '17

05 Jan '17

Greetings list, I am forking a call to multiple destinations and want to keep the only branch which sends quicker first 180/183 reply and cancel the remaining branches. Below is my related cfg snippet. modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "cancel_b_method", 2) ... route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); append_branch(); append_branch(); route(RELAY); exit; } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { xlog("L_INFO","Received $rs (IP:$si:$sp)\n"); if(status=~"18[03]"){ t_cancel_branches("others"); xlog("L_INFO","cancelled all other branches\n"); } route(NATMANAGE); } } ... But I see kamailio does not instantly send CANCEL to other branches after it has received 180/180 from any branch. Kamailio is sending CANCEL to other branches after they start sending 180/183 one by one. How can I cancel all other branches instantly? I am sure there is something wrongly configured in my configuration. Any suggestion is much appreciated. Best Regards.

3 3

location
by Slava Bendersky 05 Jan '17

05 Jan '17

Hello Everyone, Why save location will ignore on pv var when trying save on reply 200 OK. if(is_method("REGISTER") && fnmatch("$rs", "200")) { xlog("L_INFO", "OnReply: Registration is $rs OK saving --> $avp(rcvcontact) in db\n"); $var(oar) = "sip:" + $tU + "@" + $(avp(rcvcontact){uri.host}) + ":" + $(avp(rcvcontact){uri.port}); if(!save("location", "0x04", "$var(oar)")) { xlog("L_INFO", "Error saving the location\n"); } Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=619 a=26 n=xlog Dec 26 11:43:20 cavprx00 kamailio: 4(5546) INFO: <script>: OnReply: Registration is 200 OK saving --> sip:4300@client_pub_ip:31376;transport=tls in db Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=620 a=63 n=assign Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=624 a=16 n=if Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=621 a=27 n=save Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-1] (1 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-2] (2 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-2 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1004]: db_update_ucontact_ruid(): ruid:uloc-5860b463-10c5-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1133]: db_update_ucontact_ruid(): contact:sip:4300@10.18.130.27:5060;transport=udp ----> LAN side contact Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 10 chars, out: 10 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 21 chars, out: 21 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 33 chars, out: 33 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 40 chars, out: 40 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 20 chars, out: 20 chars Slava.

2 7

Kamailio v5.0 - development frozen
by Daniel-Constantin Mierla 05 Jan '17

05 Jan '17

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 5.0.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 5.0 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to reach a stable state before releasing 5.0.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Besides the new features, there were two major changes for 5.0: 1) source code tree restructuring - this should not affect the stability of the code, only installation scripts or packaging may still need tuning 2) mi (management/control interface) code has been removed. SIP routing code should not be affected by this change that much, but testing of RPC commands needs a special care. There are few RPC commands not ported yet from the MI code, they can be done during the testing period Moreover, help with updating the wiki page for migration from 4.4 to 5.0 as well as what is new in 5.0 is very appreciated. I will try post updates about them very soon. Many thanks to everyone involved in development of 5.0 and the early testers that played with master branch so far. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

unexpected char [#015]
by AL S 04 Jan '17

04 Jan '17

Hi, Kamailio is complaining about unexpected char [#015] in TO header of INVITE: Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/parse_to.c:626]: ERROR: parse_to : unexpected char [#015] in status 6: <<<sip:2057030997@72.31.12.161;tag=gK006cb08b>> . Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: <core> [parser/msg_parser.c:180]: ERROR: get_hdr_field: bad to header Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: INFO: <core> [parser/msg_parser.c:353]: ERROR: bad header field [To: <sip:2057030997@] Jan 3 14:55:41 tpa-ar-openser01 /usr/sbin/kamailio[4782]: ERROR: tm [t_lookup.c:1118]: ERROR: reply cannot be parsed Here is the TO header for the same call in both text and hex format. I am unable to see any ^M char in this TO header. To: <sip:2057030997@72.31.12.161;tag=gK006cb08b 546f3a203c7369703a323035373033303939374037322e33312e31322e3136313e3b7461673d674b30303663623038620d0a Any help would be appreciated. AS,

2 2

IMPORTANT: MI code removed (or disabled)
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, as of now, no active MI reladed code should be in master branch -- the internal library kmi and the mi_datagram, mi_fifo, mi_xmlrpc, mi_rpc and pua_mi modules were removed. Couple of modules still have some mi code disabled with ifdefs, they are pending the port to RPC commands. These are: - carrierroute - ims_dialog - mohqueue - p_usrloc - pdb - rtpengine - userblacklist - utils They should be updated in the next few days. With MI code removed, the code is slimmer and the development effort is reduced -- in the recent past there were various issue rising due to usage of the old line-based MI commands. >From now on, the RPC interface has to be used for interacting with kamailio at runtime. The tools: - kamctl - was already migrated to use RPC via jsonrpcs module. - kamcmd - was developed to use only RPC interface via ctl module. - kamcli - is also working with RPC via jsonrpcs module Helping with testing the RPC commands is very appreciated -- open an issue on bug tracker whenever you discover a problem. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Re: [SR-Users] sr-users Digest, Vol 140, Issue 3
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello Vicente, Thanks for your response, In response to your query, Yes i have Built ndb_redis module and i can see ndb_redis.so in /usr/local/lib64/kamailio/modules folder, I have installed hiredis depenency also, I have also modified the kamailio.cfg file loadmodule and modparam list, I feel like the way i am using the redis_cmd is wrong, I think i need to add it in route section of kamailio.cfg file, Can you please guide me regarding how to use redis db in order to update user information to DB on registration of each user. Thanks & Regards Hemanth Message: 8 Date: Wed, 4 Jan 2017 14:41:42 +0530 (IST) From: "hemanth(a)advaitamtech.com" <hemanth(a)advaitamtech.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <1483521102.18025940(a)apps.rackspace.com> Content-Type: text/plain;charset=UTF-8 Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth ------------------------------ Message: 9 Date: Wed, 4 Jan 2017 10:50:05 +0100 From: Vicente Hernando <vhernando(a)systemonenoc.com> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Regarding usage of Redis Module in Kamailio core Message-ID: <b003d2aa-ddd0-574d-b9e3-9b91e44976f1(a)systemonenoc.com> Content-Type: text/plain; charset=utf-8; format=flowed Hello Hemanth, just a quick response, I would check if you compiled ndb_redis module correctly (also if you have hiredis library dependency). Also I would check if you loaded ndb_redis module in your configuration file. Regards, Vicente. On 01/04/2017 10:11 AM, hemanth(a)advaitamtech.com wrote: > Hello, > > I am using Kamailio as a SIP proxy, > > I want to store Registered user information in Redis DB, > > I am able to configure REDIS DB, But i am not able to run the redis command from > > Kamailio core using the function > > "redis_cmd("srvN","SET key 1", "r")" > > I am getting undefined reference error as below > > "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" > > can anyone please help me to fix this issue > > Thanks & Regards > Hemanth > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 140, Issue 3 ****************************************

1 0

Regarding usage of Redis Module in Kamailio core
by hemanth＠advaitamtech.com 04 Jan '17

04 Jan '17

Hello, I am using Kamailio as a SIP proxy, I want to store Registered user information in Redis DB, I am able to configure REDIS DB, But i am not able to run the redis command from Kamailio core using the function "redis_cmd("srvN","SET key 1", "r")" I am getting undefined reference error as below "/usr/local/src/kamailio-4.4/kamailio/receive.c:208: undefined reference to `redis_cmd'" can anyone please help me to fix this issue Thanks & Regards Hemanth

2 1

Update: removing MI code
by Daniel-Constantin Mierla 04 Jan '17

04 Jan '17

Hello, based on lib linking in Makefiles, the following modules still use MI, categorized by their state from my point of view. Hopefully some other people will help with those listed in 2) to 6). 1) to be removed - mi_datagram - mi_fifo - mi_rpc - mi_xmlrpc - pua_mi 2) rpc commands implemented - dialog - others modules depend on some mi callback from it (qos and sst) - qos - this looks ready to wipe out the mi code, but didn't get the time not analyze properly as I don't use it usually 3) some rpc commands implemented and the rest should not be very complex - carrierroute - I implemented the dump rpc command, but it was reported that it has issues, so it needs follow up. 4) rpc commands have to be implemented, expecting not to be very complex - cplc - imc 5) expecting some degree of complexity, but they are important modules - rtpengine - rtpproxy 6) not familiar with the mi commands in these modules, so not able to assert what to expect - ims_dialog - some rpc commands are implemented, not sure if the rest of MI are used/usefull - mohqueue - it doesn't seem complex to implement rpc commands at first sight, but the indentation style didn't allowed a fast analyze on a quick look - p_usrloc - several mi commands - sst - uses some callback for MI from dialog module. qos has something similar with rpc alternative already implemented - userblacklist - several mi commands - utils - there are few mi commands related to some forwarding rules: https://www.kamailio.org/docs/modules/devel/modules/utils.html#idp21741924 The 1) to 5) should be done in a way or another, before of after freeze of 5.0.0. But 6) would require the devs of the modules (or the MI parts) to help if they want those commands via RPC. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Kamailio as loadbalancer + NAT pinning
by Vladyslav Zakhozhai 03 Jan '17

03 Jan '17

Hi, I want to use Kamailio as SIP loadbalancer and NAT helper in front of freeswitch boxes. Main tasks here are: 1. load balancing SIP requests to freeswitch boxes 2. fixing contact information during registrations and invites for nat'ed clients (kamailio is not registrar) 3. fixing sdp (or proxying rtp via rtpproxy) for nat'ed clients 4. security (check SIP messages, pike, etc) 5. outbound proxy for freeswitch boxes The main question here is about nat pinning. Acctording to module doc ( http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html) I need nathelper module and usrloc module. So I can proxy REGISTER requests to freeswitch boxes and in onreply_route check for REGISTER statuses: 1. if no success we use pike to limit bruteforce attacks 2. if success I need to save contact to usrloc db table But there is one issue. Multiple registration are allowed on freeswitch boxes and in replies (freeswitch -> kamailio) I see more than one Contact header. Is it an issue or kamailio will handle it without problems? Or I need to save Contact header from request to hash table and retreive it in onreply_route? But I do not know how to save it in usrloc db table then. I thin this is more sophisticated approach. I appreciate your advice. Thank you in advance. -- С уважением, Владислав Захожай

2 4

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 03 Jan '17

03 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

2 11

Rewrite From domain
by Gonzalo Gasca Meza 03 Jan '17

03 Jan '17

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+18888888888" <sip:+18888888888@sip.sp1.com> to From: "+18888888888" <sip:+18888888888@*sip.sp2.com <http://sip.sp2.com>*> *Call flow:* Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. ( sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like to rewrite the From domain field in this conditions: a) Calls comes from "sip.sp1.com" AND b) Call is being routed to PhoneB. Right now Im using the following code to find user and send call to B. #!ifdef WITH_ALIASDB # search in DB-based aliases xlog("L_INFO","alias_db_lookup: Call received. $rU\n"); if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif I found this in documentation: $fd - From URI domain if($hdr(From)=~"sip.sp1\.com") { ...} But not sure where is the best place to overwrite the From URI domain header. Thanks

3 7

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 02 Jan '17

02 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

1 0

Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Happy 2017!
by Daniel-Constantin Mierla 01 Jan '17

01 Jan '17

A fresh year ahead, let it be a great one for all Kamailio friends! Wishing a prosperous and safe 2017 everywhere! Happy New Year! -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

TOPOS contact issue on RE-INVITE
by Daren FERREIRA 30 Dec '16

30 Dec '16

Hello, I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations. I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary). My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header. Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers? PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | ————————— 20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 246 v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0 20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 264 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes 20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE 20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> 20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X <sip:btpsh-58656af1-769f-1@X.X.X.X> SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0 20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Length: 0 20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes And so on until timeout and BYE

3 4

Astersik kamailio integration.
by Deep Pratap Singh Tomar 29 Dec '16

29 Dec '16

Hi I have modified the file so that if i send register request to kamailio it sud register on asterisk also. i have shared you the code but i cant see anyting happening in the asterisk server. please help me ASAP. L&T Technology Services Ltd www.LntTechservices.com<http://www.lnttechservices.com/> This Email may contain confidential or privileged information for the intended recipient (s). If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system.

2 1

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

String split function
by Grant Bagdasarian 29 Dec '16

29 Dec '16

Hi, Is there a string split function built into Kamailio for use in the cfg script? I'd like to split a string of key/value pairs inside the cfg script. For example: Key1=Value1;Key2=Value2; First split on ; Then on = I think I can use the String Transformation {s.select,index,separator}, but I was hoping there is a built in function. {param.valueat,index[, delimiter]} may also be an option. Regards, Grant

2 2

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I am attaching latest config along with kamailio log file and client trace. Please let me know how I can get Kamailio to advertise public ip in record route. I followed all the steps mentioned here - http://www.fredposner.com/1457/kamailio-behind-nat/ Thanks Pranathi From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 5:09 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Cc: Mick McGrellis <MMcGrellis(a)cyracom.com>; 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: Kamailio behind NAT, ACK to private IP not advertised public IP. I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 28 Dec '16

28 Dec '16

Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 28 Dec '16

28 Dec '16

Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

1 0

rtpengine destroys voice branch on multiple branch call
by Yuriy Gorlichenko 27 Dec '16

27 Dec '16

Hi. I have 2 endpoints 1001 and 1002 1001 registered from 1 device 1002 registered from 2 devices at the same time When I called to 1002 kamailio makes 2 branches rtpengine_manage command called from branch rout for handling every branch directly (it can be different endoint types (ws/tls/udp) for each branch) When i picking up at the 1002 on one device server sends CANCEL (answered elswhere) to another devices of 1002 rtpengine_manage reacts at the CANCEL and deletes this brach after 30 seconds My question is: What is solution for use rtpengine_manage and call branches together?

1 0

set_body_multipart and append_body_part
by Diego Nadares 26 Dec '16

26 Dec '16

Hi guys, I'm trying to append a content type to a message that has only one content type. I tried using just set_body_multipart and now I'm using set_body multipart and append_body_part with no success too. The result is a message with multipart but has "two" sdp and no gtd. It has the old sdp with no ip changed for rtpproxy and the new sdp with ip changed BUT with no content type. And the message has not the end boundary (--unique-boundary-1.) Kamailio log has an info and a warning Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio.cfg ...... route[RELAY]{ $avp(ch) = 1 .... if ($avp(ch)){ set_body_multipart(); msg_apply_changes(); $var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n"; append_body_part("$var(gtd)", "application/gtd", "signal;handling=optional"); } .... } Ngrep U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060 ... Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060 .... Content-Type: multipart/mixed;boundary="unique-boundary-1". Mime-Version: 1.0. Remote-Party-ID: <sip:1152565064@172.16.213.38:5060 >;party=calling;privacy=off;screen=no. . --unique-boundary-1. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. . --unique-boundary-1. v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 58724 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=sendrecv. a=rtcp:58725. a=ice-ufrag:hhXDwrco. a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU. Any help will be very very appreciated! Thanks in advance. Diego

2 4

help configure dispatcher in Amazon ec2
by huan nguyen duy 26 Dec '16

26 Dec '16

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —————> | Load balancer —————> | Sip Server huancomputer | private IP: 172.31.11.35 | 172.31.13.113 soft phone 2(Linphone) —————> | public IP: 35.166.179.244 | 35.166.152.104 huannguyen | | The dispatcher.lst config file in /usr/local/etc/kamailio as below: # gatewayss 1 sip:172.31.13.113:5060;transport=udp <sip:172.31.13.113:5060;transport=udp> 0 0 abc=xyz User records subscriber table: ——————————————————————————————————————————————————— id | username | domain ——————————————————————————————————————————————————— 70 | nguyenhuan | nguyenhuan(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> 71 | huancomputer | huancomputer(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> Location records from location table: ——————————————————————————————————————————————————— username | domain | contact ——————————————————————————————————————————————————— huancomputer | NULL | sip: huancomputer(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:64166;transport=udp nguyenhuan | NULL | sip: nguyenhuan(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:33267;transport=udp But soft phone 1 cannot make a call to soft phone 2, below is data is extracted from wireshark: Load balancer: -------------------------------- No. Time Source Destination Protocol Length Info 177 4.463129 14.161.41.70 172.31.11.35 SIP/SDP 1433 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 179 4.463367 172.31.11.35 14.161.41.70 SIP 452 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 180 4.463421 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1215 32.552942 14.161.41.70 172.31.11.35 SIP 1105 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1216 32.553170 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1219 32.555588 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1220 32.555706 172.31.11.35 14.161.41.70 SIP 593 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1609 42.603150 14.161.41.70 172.31.11.35 SIP 1113 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1610 42.603364 172.31.11.35 172.31.13.113 SIP 1269 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1613 42.605800 172.31.13.113 172.31.11.35 SIP 749 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1614 42.605932 172.31.11.35 14.161.41.70 SIP 599 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1688 44.425237 172.31.11.35 14.161.41.70 SIP 471 Status: 408 Request Timeout | No. Time Source Destination Protocol Length Info 1689 44.425285 172.31.11.35 172.31.13.113 SIP 491 Request: CANCEL sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1691 44.426018 172.31.13.113 172.31.11.35 SIP 519 Status: 200 canceling | Data from Sip Server: ------------------------------- No. Time Source Destination Protocol Length Info 237 6.242928 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 243 6.244409 172.31.13.113 172.31.11.35 SIP 602 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 244 6.244541 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 288 6.710414 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 328 7.710454 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 396 9.710411 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 547 13.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 688 17.710406 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 835 21.710381 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 985 25.710385 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1130 29.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1279 33.710400 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1303 34.332395 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1312 34.334415 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1459 37.710402 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1609 41.710395 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | Huan Nguyen, --------------------------------------- Mobile app developer Phan Thi Company Sai Gon, VietNam Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 1

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '16

24 Dec '16

While waiting for Santa, it's a good time to also express my thanks and greetings to all the friends, developers and community members that made 2016 another amazing year for Kamailio project. Enjoy the winter holidays together with your beloved ones! Merry Christmas! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Merry Christmas to all Kamailians
by Olle E. Johansson 24 Dec '16

24 Dec '16

Friends, Trying to calm down and prepare for Christmas with the family. It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community. We’ve made great releases, had a great conference and overall done good stuff :-) For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays! A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend working with the code, helping out in discussions on the lists and in the bug tracker. And to all of you in the community - Thank You for being part of this! Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute shopping is just about to start... Keep your Kamailio running! /Olle PS. If you are not currently following the project on Twitter - it’s time! Find us as @kamailioproject

3 2

baresip version 0.5.0 released
by Alfred E. Heggestad 23 Dec '16

23 Dec '16

Hi, my apologies for the off-topic posting but I think this could be interesting for the SIP Community. Baresip version 0.5.0 is now released, you can find the announcement email here: http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html Merry Xmas from Norway! /alfred

1 0

Re: [SR-Users] rtjson + mhomed and record route (corrected)
by Diego Nadares 23 Dec '16

23 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header? ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 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; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } .... Thanks in advance. Diego 2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>: > Hi list, > > We have two eth, one with a private ip and the other with a public ip. We > have MHOMED configured. The call comes from private network and then is > routed to a public network via rtjson routing. The is routed but the thing > is that the record route is being set with the private one (It's the same > when calls come from public and is routed to a private network, record > route has the public and not the private). > > With mhomed configured shouldn't Record Route be updated as VIA header > > ".... When activated, sip-router will select a socket that can reach the > destination (to be able to connect to the remote address). (sip-router > opens a UDP socket to the destination, then it retrieves the local IP which > was assigned by the operating system to the new UDP socket. Then this > socket will be closed and the retrieved IP address will be used as IP > address in the Via/Record-Route headers)..." > > This is part of ngrep > > U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 > INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. > To: <sip:1111111111@172.16.213.38:5060>. > Call-ID: 1-13024(a)172.16.213.21. > CSeq: 1 INVITE. > Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. > .... > > U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > .... > > U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 > INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. > *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* > Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f > Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 > > This is my cfg > > mhomed=1 > > 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; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # handle retransmissions > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) { > record_route(); > } > > > Thanks in advance. > > Diego. > >

1 1

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

2 1

rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 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; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } Thanks in advance. Diego.

1 0

Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

2 3

kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 0

kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information.

1 0

Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

2 1

register_timer cb function does not have access to shared memoery var
by Andy 22 Dec '16

22 Dec '16

in mod_init I have something like list = (struct list_t*) shm_malloc ..... this list is available from any functions in the module, now there is a timer registered, in the same module register_timer(callback_f, 0 , 5) the callback_f function when tries to use list, always sees as empty -- View this message in context: http://sip-router.1086192.n5.nabble.com/register-timer-cb-function-does-not… Sent from the Users mailing list archive at Nabble.com.

2 2

new URI [] shorter than old URI - error
by Nelson Migliaro 21 Dec '16

21 Dec '16

Hello, My SIP vendor request me to replace the from in order to authenticate calls. I use: uac_replace_from("111111111", "sip:111111111@vendor-domain.com"); And when finishing calls I have this error: ERROR: uac [replace.c:607]: restore_uri(): new URI [] shorter than old URI [sip:34912345678@vendor-public-ip] Information ------------------------------------------------------------------ 34912345678 -> original CLI 111111111 -> my vendor contract vendor-domain.com -> domain of the vendor ------------------------------------------------------------------ I use the function uac_replace_from in the request_route My Kamailio version is: 4.2.8-1.1 Thank you Nelson.-

1 0

kamailio 4.4 segmentation fault when dp_translate match dialplan with ftmatch operator and empty match expression
by Julia Boudniatsky 21 Dec '16

21 Dec '16

version: kamailio 4.4.4 id:852c47 ENTRY: { PRIO: 16765956 MATCHOP: 2 *MATCHEXP: * MATCHLEN: 0 SUBSTEXP: REPLEXP: ATTRS: 1 } Please see generated core file in extension. Thank you, Julia.

2 7

nathelper - variable natping_interval per registration
by Phil Lavin 20 Dec '16

20 Dec '16

Hi all, Is it possible to set the natping_interval to a different value, per registration, at the time the user registers? For example, user A has 10 seconds and user B has 60 seconds. Thanks

3 3

auth result -3
by Jon Bonilla (Manwe) 19 Dec '16

19 Dec '16

Hi all I'm getting an auth challenge result of -3 and I don't know what it means as it's not listed in the auth module doc. Here's syslog' output: db_postgres [km_dbase.c:230]: db_postgres_submit_query(): sending query ok: 0x7f36d1f75f00 (2) - [select secret from sip_buddies where username='48481628780'] db_postgres [km_res.c:108]: db_postgres_get_columns(): 1 columns returned from the query db_postgres [km_res.c:133]: db_postgres_get_columns(): RES_NAMES(0x7f36d1f92238)[0]=[secret] db_postgres [km_res.c:166]: db_postgres_get_columns(): use DB1_STRING result type db_postgres [km_res.c:211]: db_postgres_convert_rows(): no rows returned from the query auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@' INFO: <script>: Ha fallado la auth con resultado -3 cheers Jon

3 4

INFO: jsonrpcs used instead of mi_fifo in the default kamailio.cfg and kamctl
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, with MI code planned to be removed, I updated the kamctl and default kamailio.cfg to use jsonrpcs (jsonrpc server) module instead of mi_fifo to interact with kamailio at runtime. The default fifo file for rpc command is /var/run/kamailio/kamailio_rpc.fifo and expects JSONRPC commands as described at: - http://kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idp31525708 Switching back to mi_fifo is still possible by loading the module and editing kamctlrc, but the module might not make it to 5.0, so for new installations and testing is better not to use it. Few rpc commands are missing in carrierroute, otherwise the kamctl commands should work just as before, but be aware that the output is now in json format. Testing of the master branch with the latest kamctl and providing feedback will be really appreciated -- I could do tests only for the modules I typically use. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Forking with 404 replies
by José Seabra 19 Dec '16

19 Dec '16

Hello there, I have a Kamailio server acting as registrar server and when the subscriber has more than 1 AOR, kamailio sends an INVITES to each AOR, if one of them replies 404 what should be the kamailio behavior: 1. should Kamailio reply with an ACK and not pass the 404 to UAC? 2. should Kamailio forward the 404 to UAC? I'm asking this because i'm facing the second behavior, that is, kamailio is forwarding the 404 to UAC and the call is terminated. -- Regards José Seabra

2 3

RFC: renaming the modules with - (dash) in name
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, given the work for 5.0 which will bring major changes on source tree structure, I am considering to remove the - (dash) from the modules that have it in the name. There are 4 of them now: cpl-c janssonrpc-c jsonrpc-c jsonrpc-s One of the reasons is that this kind of name doesn't work for the feature of defining MOD_modname for each loaded module, because - is a separator for ids in config language grammar, so for example, MOD_jsonrpc-c cannot be used in an #!ifdef. More over, for embedded interpreters, most of exported functions there use kamailio module name as package name, and again this will break. Of course, a different package name can be defined, but won't be coherent anymore. Anyone against this renaming? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Re: [SR-Users] Question on multiple 200OK received after forking
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, well, the specs say that all 200ok must be forwarded to the caller, because a middle point cannot decide for the user making the call. Even sending an ACK for the 200ok from a middle point is not enough because the callee of that branch will think it is in an active call. The caller can take different actions, the usual is to ACK the 2nd 200ok and then immediately send BYE. But it can also decide to end previous branch and switch to this one (e.g., uses better codecs, or has video, etc...). Cheers, Daniel On 14/12/2016 15:05, Marino Mileti wrote: > > Hi guys, > > > > I've a Kamailio instance listening on both eth0 and wlan0. > > Kamailio forks call to two branches, so both clients rings together. > > > > In case (very very very rare J) of simultaneous answers is there any > way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to > arrange a semaphore with lock/unlock capabilities of cfgutils, I've > used the drop command but client still sending the 200OK because it's > waiting for an ACK L > > > > Any help will be very appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it/ <mailto:marino.mileti@alice.it>// > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

(Correction) unusual characters added to reINVITE message
by AL S 19 Dec '16

19 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

2 1

Limiting calls based on CLI and dst number
by Cibin Paul 19 Dec '16

19 Dec '16

Hi, Is there a way to limit the number of active calls per cli or dst number basis? Yesterday my system had a ddos attack in which 123456 (CLI) was hitting 2345 (DID) at a rate of 100-200 calls per minute. I would like to restrict to 2-4 simultaneous calls per CLI/DST. Any help would be appreciated. Thanks, Cibin

4 9

http_async_client - queries
by Ramachandran, Agalya (Contractor) 17 Dec '16

17 Dec '16

Hi team, We are trying to use kamailio for our project purpose, which will be used as a proxy server. Am trying to use "http_async_client" module to send my rest queries asynchronously. I have few queries regarding the usage of http_async_query() and using PUT method. http_append_header("Authorization: Bearer $var(auth)"); http_append_header("Content-Type: application/json"); http_set_method("PUT"); http_async_query("http://sdk-evm-wcdc-c-001.rtc.sys.comcast.net/events/createxmpprooteventfor…", "{'from_tn':'$fU','to_tn':'$tU','callid':'ci'}", "HTTP_REPLY"); Here am passing from, to, callid in post data field. Passing as 'key':'value' is mandatory or I can pass only value- ? In the file, async_http.c - in async_send_query(), if I modify the post data to the format I need, I see the value is truncated up to three fields only. How can I achieve that my payload format whatever I form in async_send_query(),doesn't get truncated and will be passed to the server-? Can you please explain how this works-? Also, I observed that kamailio is restricting the content-length of HTTP request to 256. Is this something modifiable-? If so please guide me how this can be achieved. Regards, Agalya

2 2

sipp dead calls
by Diego Nadares 16 Dec '16

16 Dec '16

Hi guys, We are testing kamailio with sipp. We are running it with 20cps and some calls do the following. When Kamailio is processing the 'ringing' a '200ok' arrives in the middle. First, kamailio forwards the 200 ok and then the ringing. 'ACK' arrives and, I suppose, the call is established. The thing is than when the 'BYE' arrives kamailio responds with a 404. This is the summary of the call Id Time Source Destination Protocol Len Info 6483 2016-12-14 11:26:06.264697 SIPP-A KAMAILIO SIP/SDP 692 Request: INVITE sip:11111111@172.16.213.38:5060 | 6484 2016-12-14 11:26:06.264937 KAMAILIO SIPP-A SIP 367 Status: 100 trying -- your call is important to us | 6485 2016-12-14 11:26:06.266327 KAMAILIO SIPP-B SIP/SDP 1179 Request: INVITE sip:22222222@172.16.213.31:5060 | *6486 2016-12-14 11:26:06.267217 SIPP-B KAMAILIO SIP 566 Status: 180 Ringing | * 6487 2016-12-14 11:26:06.267268 SIPP-B KAMAILIO SIP/SDP 733 Status: 200 OK | 6488 2016-12-14 11:26:06.267758 KAMAILIO SIPP-A SIP/SDP 788 Status: 200 OK | *6489 2016-12-14 11:26:06.267833 *KAMAILIO SIPP-A* SIP 442 Status: 180 Ringing | * 6490 2016-12-14 11:26:06.268868 SIPP-A KAMAILIO SIP 493 Request: ACK sip:127.0.0.8;line=sr-N6IAzBFwMJZfWJZLM.M7MlF-W.y6Mx14NEt7Nw05NhPQKjaP | 6491 2016-12-14 11:26:06.269162 KAMAILIO SIPP-B SIP 609 Request: ACK sip:172.16.213.31:5060;transport=UDP | 6492 2016-12-14 11:26:06.269614 SIPP-A KAMAILIO SIP 404 Request: BYE sip:11111111@172.16.213.38:5060 | 6493 2016-12-14 11:26:06.269782 KAMAILIO SIPP-A SIP 348 Status: *404 Not here* | We are using modules rtjson, evapi, uac, topoh, rtpproxy for all calls. My debug level is -1. With higher levels this behavior increase. Kamailio is running in a virtual machine with centos7 with 8 cores and 8gb of ram. Do you need any further information? I can send you a pcap or ngrep file. Best regards, Diego.

2 3

unusual characters added to reINVITE message by Kamailio
by AL S 16 Dec '16

16 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

1 0

ACK not routed to destination
by Wim Van Cauwenbergh 16 Dec '16

16 Dec '16

Hi, I have setup kamailio with the dispatcher module and have a problem with the ACK in response to 200OK not being forwarded by kamailio: Scenario is very simple: sipp(.219) --> kamailio (.220) --> sipp(.239) SIP trace: root@DNS-enum-8Gb-3:/var/log# ngrep -W byline -d eth0 port 5060 interface: eth0 (10.57.26.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 10.57.26.219:5060 -> 10.57.26.220:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.239:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Record-Route: <sip:10.57.26.220;lr>. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.219:5060 -> 10.57.26.220:5060 ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 ACK. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. XContactinOK: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # *U 10.57.26.219:5060 <http://10.57.26.219:5060> -> 10.57.26.220:5060 <http://10.57.26.220:5060>* *ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0.* *Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5.* *From: sipp <sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>>;tag=15698SIPpTag021.* *To: sut <sip:service@10.57.26.220:5060 <http://sip:service@10.57.26.220:5060>>;tag=14783SIPpTag018.* *Call-ID: 1-15698(a)10.57.26.219 <1-15698(a)10.57.26.219>.* *CSeq: 1 ACK.* *Contact: sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>.* *Max-Forwards: 70.* *Subject: Performance Test.* *XContactinOK: <sip:10.57.26.239:5060;transport=UDP>.* *Content-Length: 0.* . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@10.57.26.219:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. *kamailio.conf:* # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", # "mysql://openser:openserro@localhost/openser") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; } # 1= only check the header RURI, 1126 checks some other general stuff if(!sanity_check("1126", "1")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing xlog("#######WITHINDLG:hastotag $si:$sp\n"); if (loose_route()) { xlog("#######WITHINDLG:looseroute $si:$sp\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("#######WITHINDLG:ACK transaction found $si:$sp\n"); # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. xlog("#######WITHINDLG:ACK no transaction found so discard $si:$sp\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Debug: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:10.57.26.239:5060 ;transport=UDP> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-15698-1-5>; state=16 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=14783SIPpTag018 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [57]; uri=[sip:service@10.57.26.220:5060] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [sut < sip:service@10.57.26.220:5060>] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: siputils [checks.c:106]: has_totag(): totag found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:hastotag 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=8 T start=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=53919, isACK=1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=9 T end=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:ACK no transaction found so discard 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)

3 3

Siremis installation with postgresql.
by José Seabra 16 Dec '16

16 Dec '16

Hi there, I'm facing some issues trying to install siremis with postgresql. The file Config.xml has the correct configuration for postgresql driver but It seems that during installation phase siremis can't create database structure for postgresql. Anyone have installed siremis with postgresql, if yes can you share the steps needed for this installation. -- Cumprimentos José Seabra

4 6

kamcli - transferring the github repository to kamailio organization
by Daniel-Constantin Mierla 16 Dec '16

16 Dec '16

Hello, kamcli (kamailio command line interface tool written in python) got a consistent set of features trying to become in pair with kamctl. It is currently hosted by asipto organization on github, but I am considering to transfer (move) it to kamailio organization in order to manage easier the access for kamailio devs as well as increase its exposure to community. It interacts with kamailio using rpc commands via jsonrpcs module. You can read more about what it can do at: - https://github.com/asipto/kamcli The commands are implemented in a plugin-like fashion -- see the ones done so far: - https://github.com/asipto/kamcli/tree/master/kamcli/commands Given that it can handle better the jsonrpc responses, provides more options to format the output and the help is embedded in each command, I think it is a good candidate to replace the old shell(bash)-based kamctl. Probably we can adjust kamctl for rpc when mi code will be removed, but the options to build and extend in long term a good cli for kamailio are quite limited in shell. For the sake of an example, here is the implementation of 'ps' command, handling the json rpc to display a very compact list of kamailio processes: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_ps.py Dealing with database records is also rather trivial: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_group.py Anyone with comments/suggestions/alternatives? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Forwarding Emergency calls based on caller IP address
by João Vitor Torres 15 Dec '16

15 Dec '16

Hello all, I am looking for a solution to forward emergency calls inside an enterprise network. Is there any way to forward calls based on the IP address of caller device? The requirement is that calls to a unique emergency number must be forwarded to specific local numbers in each building, and the IP address of the caller is the criteria for choosing the destination. A table maps IP nets to specific local numbers. The SIP contact header has the caller IP address, but I am new to Kamailio and I am not sure where to start. Any suggestions? Thanks in advance. João Vitor.

2 1

Re: [SR-Users] htable lock - unlock
by Daniel-Constantin Mierla 15 Dec '16

15 Dec '16

Hello, the log message you refer to is inside the if block: if ($sht(pht=>$ci)!=$null) { ... } but you don't set the item in memory, no assignment like: $sht(pht=>$ci) = ...; The sht_lock(pht=>$ci) is just locking the slot where the value of $sht(pht=>$ci) will be stored, but don't store any valye for that variable. Cheers, Daniel On 15/12/2016 11:46, Marino Mileti wrote: > > Hi guys, > > > > I'm going out of mind J > > I've this piece of script inside an "on_reply_route" that captures > only 200OK of INVITE (i would like to save in ht the contact of who > has answered a call) > > > > sht_lock("pht=>$ci"); > > if ($sht(pht=>$ci)!=$null){ > > xlog("L_INFO","Before assignment.."); > > $sht(pht=>$ci)=$var(myvar); > > xlog("L_INFO","Assigned done..."); > > sht_unlock("pht=>$ci"); > > } > > else{ > > xlog("L_INFO","Skipping.."); > > } > > sht_unlock("pht=>$(hdr(Call-ID))"); > > > > The situation is that I never see the "Assigned done" message in log > file... > > > > Is this the right use of sht_lock/sht_unlock method? > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

Updating to SIREMIS 4.3
by Kevin Greene 15 Dec '16

15 Dec '16

Hello fellow SIP people: I upgraded from SIREMIS 4.2 to 4.3 for my Kamailio system and I now get the error message shown below. Any ideas on how to solve it? I am assuming there was a change on how SIREMIS did a date query and my version of Linux (6.8) or my PHP does not support it as expected. Any help would be useful. Thanks. Kevin [{"target":"ERROR","content":"\n[2016-12-14 22:59:55 (GMT)] An exception occurred while executing this script: \nError message: #2, date(): It is not safe to rely on the system's timezone settings. You are *required* to use the date.timezone setting or the date_default_timezone_set() function. In case you used any of those methods and you are still getting this warning, you most likely misspelled the timezone identifier. We selected 'UTC' for 'UTC\/0.0\/no DST' instead<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php:160<\/font> \n *function:<\/b> errorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/sysheader.inc 117\n *function:<\/b> userErrorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \n *function:<\/b> date ( \"m\/d\/Y\" ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php 160\n The "date" command returns: $ date Thu Dec 15 01:13:41 UTC 2016 The /etc/localtime entry is: $ more /etc/localtime TZif2 UTC $ which php /usr/bin/php $ php -version PHP 5.3.3 (cli) (built: Aug 11 2016 13:38:15) Copyright (c) 1997-2010 The PHP Group Zend Engine v2.3.0, Copyright (c) 1998-2010 Zend Technologies PHP I am using is the latest patched version I have. ***

3 2

htable lock - unlock
by Marino Mileti 15 Dec '16

15 Dec '16

Hi guys, I'm going out of mind J I've this piece of script inside an "on_reply_route" that captures only 200OK of INVITE (i would like to save in ht the contact of who has answered a call) sht_lock("pht=>$ci"); if ($sht(pht=>$ci)!=$null){ xlog("L_INFO","Before assignment.."); $sht(pht=>$ci)=$var(myvar); xlog("L_INFO","Assigned done..."); sht_unlock("pht=>$ci"); } else{ xlog("L_INFO","Skipping.."); } sht_unlock("pht=>$(hdr(Call-ID))"); The situation is that I never see the "Assigned done" message in log file... Is this the right use of sht_lock/sht_unlock method? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Kamailio 4.3 Rewriting From and db_alias values
by Gonzalo Gasca Meza 15 Dec '16

15 Dec '16

Hi forum, This is the scenario: VM <- Kamailio -> SP1 ----- SP2 ---- PhoneA (john) PhoneB PhoneA is configured in *dbaliases* table: alias_username, alias_domain, username, domain 11111111, sp1.com, john, sp2.com When PhoneB dials 1111111, Kamailio converts 111111(a)sp1.com to john(a)sp2.com Now I need to provide VM services to PhoneA. (Opposite direction) PhoneA calls voicemail but the calling number is john(a)sp2.com I need to have SP1 use the 11111111(a)sp1.com format How can I overwrite the From? Pseudo algorithm ------------------------ If destination is VM and from domain ($fU) is sp2.com: Find user URI: ( In this case john extracted from john(a)sp2.com in dbaliases) If user is found: rewrite From URI: use alias_username@alias_domain I'm using default kamailio.cfg and calls successfully are forwarded to VM Thank you

2 1

Question on multiple 200OK received after forking
by Marino Mileti 14 Dec '16

14 Dec '16

Hi guys, I've a Kamailio instance listening on both eth0 and wlan0. Kamailio forks call to two branches, so both clients rings together. In case (very very very rare J) of simultaneous answers is there any way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to arrange a semaphore with lock/unlock capabilities of cfgutils, I've used the drop command but client still sending the 200OK because it's waiting for an ACK L Any help will be very appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

question on call cancel
by Kelvin Chua 14 Dec '16

14 Dec '16

I have a peculiar scenario, maybe you guys have an easy solution: 1. a call comes in through an SBC (which is also kamailio) 2. SBC passes call to core kamailio server 3. call reaches 180 ringing state 4. client suddenly changes from wifi to 4G (using 4G IP, VIA headers are all updated) 5. client sends cancel and receives cancelling msg from SBC 6. SBC sends cancel to kamailio core 7. kamailio core sends back Request Terminated after termination the issue is, this message was sent to the wifi IP not the 4G when checking the network dumps, looks like this was because the INVITE was received on kamailio core from the old wifi IP. Is there maybe a way to update the transaction perhaps? Kelvin Chua

2 2

Multi-Domain dispatcher
by Satish Patel 14 Dec '16

14 Dec '16

We have currently dispatcher running for single domain, but in future we have more domain coming so i want to do multi domain dispatching for example dispatcher redirect request for foo.com to foo.registar and bar.com to bar.registar SIP server currently we have following code in dispatcher logic. what should we do to handle multi-domain if(!ds_select_dst("1", "2")) { # if we are here that means no destination is available. We notify the user by 404 and exit the script. xlog("L_NOTICE", "No destination available!"); send_reply("404", "No destination"); exit; }

2 1

regular expression question
by Satish Patel 14 Dec '16

14 Dec '16

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

3 3

event for new tcp connection from client
by Andy 14 Dec '16

14 Dec '16

similar to tcp closed, can this be achieved when a client is connect to tcp -- View this message in context: http://sip-router.1086192.n5.nabble.com/event-for-new-tcp-connection-from-c… Sent from the Users mailing list archive at Nabble.com.

3 6

Kamailio TM - t_set_fr
by José Seabra 13 Dec '16

13 Dec '16

Hello there, I'm using t_set_fr(0,2000) on calls to registered phones but by any reason this isn't working, the call only enters on the failure route after 30 seconds in case of no response from SBC/Phone, anyone has any idea what can be the reason for that? if(lookup("location")) { xlog("L_INFO", "routing to registered phone ruid=$ruid R=$ru - ID=$ci \n"); t_set_fr(0, 2000); t_on_failure("ASBC_FAILURE"); t_on_branch("ASBC_BRANCH"); route(RELAY); exit; } My RELAY route: route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } version: kamailio 4.4.2 (x86_64/linux) Thanks -- Cumprimentos José Seabra

2 11

Accessing SDP changes
by Pete Kelly 13 Dec '16

13 Dec '16

Hi I am using Kamailio 4.4. and calling the function sdp_remove_transport in order to remove some SRTP media lines. The function works (of course!) however I would like to get access to the modified SDP within the cfg. I thought this would do the trick: sdp_remove_transport("RTP/SAVP"); sdp_get("$avp(foo)"); xlog("$avp(foo)"); However $avp(foo) does not contain the newly modified SDP. Is there a way for me to access it? I have been looking at things like t_save_lumps() and msg_apply_changes(). Any pointers or confirmation that this is or isn't possible would be very welcome! Thanks Pete

3 5

Tsilo iOS Push Notifications with Kamailio/Asterisk Integration
by Benjamin Fitzgerald 12 Dec '16

12 Dec '16

Hello, my goal is to address iOS 10's removal of periodic TCP connections in iOS apps with their replacement PushKit, a push-based alternative all VoIP apps should implement. In order to have iOS apps receive SIP calls in the background, I believe I need to use tsilo to suspend the INVITE transaction send a push notification to the device. I have been following the presentation here ( http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…) however I am not sure how to integrate it with Asterisk ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). My questions can be summarized below: (1) What changes need to be made from Federico's presentation to use with the Kamailio/Asterisk integration? (2) How can I modify the config to send push notifications for EVERY invite? This is to address the scenario where a device goes offline before the SIP registration expires and a call is received within that window. iOS is very aggressive about killing background applications so it's likely as soon as user leaves the VoIP app, the device will become unreachable. Thanks for any help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us

2 3

DTMF Using INFO
by Nahum Nir 12 Dec '16

12 Dec '16

Hello, I tried Googling but didn'tfind anything. Is it possible to support DTMF using INFO? Thanks, Nir

4 8

msilo timestamp
by Slava Bendersky 12 Dec '16

12 Dec '16

Hello Everyone, What format stored inc_time in msilo module ? I am trying convert it to human readable format, but time looks UTC SELECT FROM_UNIXTIME($(xavp(dbquery[$var(i)]=>inc_time)), '%M %D %Y %h:%i:%s') Slava.

2 1

Error using Siremis 4.3.0 with Kamailio 4.4 and PHP 7
by Diogenes Aviles 09 Dec '16

09 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in page "Administration-User Management". When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/ openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 I verified and the file "/var/www/siremis-4.3.0/openbiz/bin/Resource.php" exist. But I don't have any idea where the error is. Some help will be appreciate. Regards -- Diogenes

2 2

Asterisk Security Advisory (AST-2016-009)
by Matthew Jordan 09 Dec '16

09 Dec '16

Hey all - The Asterisk project just released a security advisory for a security vulnerability in which Asterisk using chan_sip with a proxy can allow for unauthenticated calls. This affects all supported versions of Asterisk (11, 13, 14). Since that may be relevant to those on this mailing list who are not also on the asterisk-users mailing list, I thought it prudent to mention it here as well. A description of the vulnerability follows: Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. The announcement can be seen here: http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.ht… Thanks again to Walter Doekes for reporting the vulnerability and providing the patch to fix it. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

3 2

IP-based security + registrations
by JLL 09 Dec '16

09 Dec '16

Hello, I would like to know if anyone can help me accomplish the following: I have a Kamailio server (4.2.3 on CentOS 7 x64) and 5 Asterisk servers (1.6.2.13 . I know, it's old but we have a lot of custom modules written for it). I need for certain IP's to be "trusted" (able to send INVITE's to K from the internet - these would be our origination providers), but any other IP's that come along would need to register before sending an INVITE (the REGISTER would be sent to any of the Asterisk servers for validation, if this is possible). Can this be done? Thank you!

2 1

Siremis manual to learn
by Saranga Nagesh 08 Dec '16

08 Dec '16

Hi, I have installed kamailio and Siremis in centos but I don't know how to use Siremis. Can you forward me manual or link to learn about Siremis. Thanks & Best Regards, Nagesh Saranga

2 1

Action: restructuring the source code tree
by Daniel-Constantin Mierla 07 Dec '16

07 Dec '16

Hello, today I am going to do the restructuring of the source code tree as discussed previously. - http://lists.sip-router.org/pipermail/sr-dev/2016-December/037715.html If anyone has patches to push using existing structure, do it before 12:00UTC today. If you need more time, write to sr-dev and let's see how we can find an easy solution. Once I finish with the restructuring, I will send another notification, from that moment the commits can go on as usual. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Send SIP Info within a dialog using $uac_req(method)="INFO"
by Jonathan Hunter 07 Dec '16

07 Dec '16

Hi Guys, Is it still the case that when using uac_req_send, you cant send withing a specific dialog?I can modify call-id, but I presume tags may be more of a problem?See old post below from 2015; > I am familiar with uac_req_send. but how do I send it with in a > specific dialog and with data in the INFO req ? sending a new request inside a dialog is not possible with uac_req_send(). It is not easy over all because you change the sequence order (CSeq value). Practically, you need to track how many requests you sent from the middle to update (and restore in reply) when caller or callee sends a new request. dialog module can track changes in CSeq for requests sent to callee, being used now for authentication of INVITE to another provider, when Kamailio adds the credentials. But for more you would need to extend the dialog module. I just need to send a SIP info within an established dialog to stop some function up stream, so wondered if this is still a blocker? Many thanks Jon

2 40

Error using Siremis 4.3.0
by Diogenes Aviles 07 Dec '16

07 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in Administration-User Management. When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 Regards -- Diogenes

2 1

Getting destination ip
by David Villasmil 06 Dec '16

06 Dec '16

Hello Guys, I'm wondering how do I get destination IP on a reply? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 7

Re: [SR-Users] Local messages are malformed in Kamailio 4.4.X
by g.Helio 06 Dec '16

06 Dec '16

2 1

Limiting Registration Contacts
by Colin Morelli 05 Dec '16

05 Dec '16

Hey all, I know Kamailio's registrar module has a max_contacts parameter that will limit the number of active contacts for an AOR. However, is there any mechanism to control what happens when that number is hit? When using a web client where every page refresh registers a new contact with Kamailio, it would be useful to automatically roll off the oldest registrations and replace to make room for the new ones. I know this can be manually accomplished with sqlops, but I was hoping something like this already existed. Best, Colin

3 5

INFO: modules purple and iptrtpproxy moved to obsolete folder
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, be aware that the modules purple and iptrtpproxy were moved to obsolete folder. They were not compiling for several years and nobody showed interest in updating them. If anyone wast to rescue them, she/he is welcome to get them back and upgrade the code to work with current versions of their dependencies. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

INFO: kcore internal lib removed
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, mainly for those having private modules or other components for kamailio, take note that the internal library kcore has been removed. It was supposed to be a termporary place where to keep some kamailio specific functions at the time of merging with ser code back in 2008. The main code was related to internal statistics, which was re-implemented as fast counters several years ago, in libkcore being only some wrappers. You have to include now directly the counters.h from core. There were several files with some str operations helpers, not they are exported via strutils.h from core. The sst parser was used only by sst module and moved to it, same being done with some stats helper functions used only by snmpstats module. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

kamailio bridge internal and external network
by zhang90501 05 Dec '16

05 Dec '16

My kamailio listen at internal ip and external ip(advertise parameter), it send message to freeswitch from internal ip and to client from external ip. How can I configure my kamailio to auto generate correct record route header?

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