- sr-users - kamailio.org

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 ****************************************

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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

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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

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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

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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

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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

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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

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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

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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

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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

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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

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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

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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

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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. > >

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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

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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

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