sr-users January 2017

sr-users@lists.kamailio.org

87 participants
111 discussions

Kamailio and default paramenters to a specific gatewy

by Laura

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

7 years, 3 months

Diameter AVP handling from kamaiio.cfg

by Zoltán Miricz

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*

7 years, 3 months

Kamailio on Cluecon Weekly – Jan 11, 2017

by Daniel-Constantin Mierla

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

7 years, 3 months

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".

by Daniel-Constantin Mierla

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

7 years, 3 months

[Error] Install Kamailio 4.4 in Mac 10.12

by Luis De Marchi

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/>

7 years, 3 months

t_precheck_trans

by Annus Fictus

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

7 years, 3 months

use case

by Thufir Hawat

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

7 years, 3 months

using kamailio to forward ipv6 to ipv4

by anfecora

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.

7 years, 3 months

why a register error?

by Olvera, Victor

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

7 years, 3 months

Re: [SR-Users] BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".

by Nihar Ranjan Deb

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

7 years, 3 months

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sr-users January 2017