- sr-users - kamailio.org

Releasing v4.2.1
by Daniel-Constantin Mierla 11 Nov '14

11 Nov '14

Hello, I am considering to release first minor version out of branch 4.2 as v4.2.1 next week, by Tuesday or Wednesday. If there is any comment regarding it, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

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ALIAS_DB replacing From and Contact fields
by Martin Schiøtz 11 Nov '14

11 Nov '14

Hi I'm using the ALIAS_DB module. username = 101 alias_username = 44444444 (fictive public phone number) 'alias_db_lookup("dbaliases")' is working fine for incoming calls from PSTN. Calling 44444444 R-URI is replaced with 101 SIP uri. But when 101 is calling to a PSTN number how do I replace the From and Contact fields with 44444444? Cheers, Martin

1 0

RTP end points
by Javier Ricke 10 Nov '14

10 Nov '14

i used asterisk, and install kamailio for traffic RTP can be send between end points, and do not have to go through RTP proxy, any work this form?, and client behind to NAT.. regards!

2 2

beginner's problem with sipping
by Petr Wozniak 10 Nov '14

10 Nov '14

Dear sr-users, I am beginner and apologize for bothering you with such a trivial problem. I would like to test sip UDP and TCP keepalive pinging. Kamailio and client are on the same network (192.168.1.44 and 192.168.1.20) that implies not behind NAT. So in kamailio.cfg I have set modparam("nathelper", "ping_nated_only", 0) My simple kamailio.cfg please you find enclosed. >From logfiles on client's side implies that neither UDP nor OPTIONS pings are sending from kamailio. If somebody would be so kind to help me I will be very glad. Thank you in advance Regards Petr

2 3

Counting xavp stack
by Alex Balashov 10 Nov '14

10 Nov '14

It would be great if there were some way to count the depth of a stack of XAVPs, e.g. $xavp(root=>num) = 1; $xavp(root[0]=>num) = 2; $xavp(root[0]=>num) = 3; ... $var(depth) = $(xavp(root=>num){avp.count}); Right now, the only way to do this that I know of is something like: $var(i) = 0; while(defined $xavp(root[$var(i)]=>num)) $var(i) = $var(i) + 1; $var(i) = $var(i) - 1; Is there a secret I don't know about? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Setting up a IMS Server Platform using Kamailio 4.2.x code
by Austin Einter 10 Nov '14

10 Nov '14

Hi All Last few days we have tried to setup Kamailio for IMS server setup. We have not been successful. Now we are planning to use Kamailio 4.2.0 code base (Latest) to setup Kamailio IMS setup. I am looking for a link/document, that gives proper step by step instructions how to setup Kamailio as IMS servers (P-CSCF, I-CSCF and S-CSCF). Can somebody kindly provide the official link or any other working link, that I can follow to setup P-CSCF, I-CSCF ,S-CSCF and HSS. Many thanks Austin

2 1

Fwd: Extension and carrier same ip problem
by Martin Snelgoedkoop 10 Nov '14

10 Nov '14

> Hi, we use Kazoo which loads acl and dial plan out a DB. > We run in to smth that seems freeswitch default behavior so that's why I write here. > > We have a carrier that needs to register a extension to our platform as well. > They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct. > > The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route. > > I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy. > > The setup is kamailio>freeswitch > > > Met vriendelijke groet, Martin > > Team snelgoedkoop

3 2

Kamilio verify online servers
by Marino Mileti 10 Nov '14

10 Nov '14

Hi guys, I've an architecture with multiple Kamailio. I would like to implement some check that allow me to verify if a server is online or not so for example i can use msilo to save chat message and relay it when the server come's up. Is it possible to use UAC module to send a REGISTER message from one server to another so i can "dump" all message in very simple way? Is there any example ? Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamilio-verify-online-servers-tp132… Sent from the Users mailing list archive at Nabble.com.

2 1

htable locking
by Alex Balashov 10 Nov '14

10 Nov '14

The documentation for the htable module gives no guidance on whether sht_lock()/unlock() should be used in all situations when writing or reading from the htable. Should it be? I was under the impression (not supported by anything said in the documentation) that read and write access to the htable is fundamentally thread-safe, i.e. it is safe to both read and write to the htable even if the potential exists for another worker process to do the same concurrently. Is that not the case? If not, should there not be strong admonitions to use sht_lock/sht_unlock around concurrent operations--and it sounds like nearly all useful applications of the htable are essentially concurrent? Thanks! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Generate REFER msg
by Agiftel 10 Nov '14

10 Nov '14

Hi all, i need some advice on how to solve a problem. I've a scenario like following: [SIP PBX] <---->CUCM (Call Manager v 10.5 ) User A: belong to CUCM User B: belong to SIP PBX User C: belong to SIP PBX User A calls User B. After some conversation, User A wants to transfer call to User C. CUCM transfer the call with re-INVITE; in this case on SIP PBX 2 channel SIP are used. CUCM does not support transfer, on Trunk, using REFER method. My idea is using Kamailio beetwen SIP-PBX and CUCM and handles call generating REFER msg in some way. Is this idea a good one? Or there is a better way? Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Generate-REFER-msg-tp132317.html Sent from the Users mailing list archive at Nabble.com.

2 1

rr problem: sips instead of sip
by Jon Bonilla (Manwe) 07 Nov '14

07 Nov '14

Hi all I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based on kamailio 4.1.6) The problem is as follows: A custom mobile client B blink for linux The path of a call from B to A would be this one: B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and registrar and sems acts as B2BUA. Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the inside. Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080). My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the Kamailio LB to the proxy. The problem is that the kamailio proxy added itself as sips in the record-route header: U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080 INVITE sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0' Record-Route: <sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->' Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Record-Route: <sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route: <sip:lb@127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>' Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0' Via: SIP/2.0/UDP 10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x' When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy: NOTICE: <script>: New request on lb - M=ACK R=sip:ngcp-lb@XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 NOTICE: <script>: Relaying request, du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--', fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1 WARNING: <core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket: protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062) I think the problem might be in the way the client registers? This is the registration info where it uses sips in the contact: contact: sips:test1@10.0.1.5:41956;rinstance=145658D8;transport=tcp If I try the same calling blink, which has the following contact in the registration it works without issues as the proxy doesn't insert sips in the Record-Route header: contact: sip:91807432@10.10.10.4:57160;transport=tls Any thoughts? cheers, Jon

3 2

Kamailio v4.0.7 Released
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, Kamailio SIP Server v4.0.7 stable release is out. This is a maintenance release of the 3rd old stable branch, 4.0, that includes fixes since release of v4.0.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.7, or even better, to 4.1.x or 4.2.x, as the focus from now on will be on maintenance of these release series. For more details about version 4.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-0-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Releasing v4.0.7
by Daniel-Constantin Mierla 07 Nov '14

07 Nov '14

Hello, I am planning to release v4.0.7 based on latest source code in branch 4.0. This branch is going out of official maintenance, the current latest stable branches being 4.2 and 4.1. If you have any remarks regarding it, write back to mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 2

cdp.so segment fault
by Tom Secula 06 Nov '14

06 Nov '14

I am just wondering if anyone has seen this type of segfault on 4.2 ? I'm trying to run kamailio as an scscf on centos 6.6 ? Nov 6 14:17:20 scscf kernel: kamailio[6246]: segfault at 0 ip 00f3f425 sp bfa0e9f4 error 6 in cdp.so[f08000+96000]

1 0

Re: [SR-Users] RPC uac.reg_info hangs second time
by Dan Christian Bogos 06 Nov '14

06 Nov '14

Hey Daniel, Thanks for your fast reply! I am using the 4.2.0 out of debian packages (not nightly). Will shift on nightly and retest. Not a big deal regarding 150 seconds, maybe it should be a good reason for async reloads (reply with the delay number of seconds when reload will happen). Reloads can be always implemented outside of kamailio but since everybody would need it I assume a more centralized approach would be useful. Anyway, thanks again! DanB On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Hello, > > are you using branch 4.2 or some tgz/package from 4.2.0 release? There > was a fix related to such issue, so 4.2 branch must be used (or debian > nightly builds for 4.2). > > I will have to check the code how flexible that is, but afaik, the limit > on reload interval was thought to avoid sending again registration > requests while retransmissions from previous one can occur. > > Cheers, > Danel

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Kamailio v4.1.7 Released
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, Kamailio SIP Server v4.1.7 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.7. For more details about version 4.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-1-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

how to check if reply is retransmission?
by Juha Heinanen 06 Nov '14

06 Nov '14

is there a means to check if reply to t_relay()ed request is retransmission? there is t_is_retr_async_reply(), but that does not apply to regular transactions. i would like to avoid double calling of rtpengine_answer(). -- juha

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AT&T and UAC cancel after 200 OK
by Yuriy Gorlichenko 06 Nov '14

06 Nov '14

Hello. I use UAC module for trunks and have trouble when call to mobile endpoint of AT&T provider. When endpoint pickup call ans 200 OK reply come into kamailio, kamailio send CANCEL. so at the endpoint I see than call is dropped and at kamailio client I hear voicemail speech form AT&T endpoint. Provider say that his trace is OK... My sip trace bellow: IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129 E.......@.l. ...6........q..INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651 E.......*.Z.6... .........}.SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth" Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410 E.......@.o. ...6.........j%ACK sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 ACK Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6..........`INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6.........._INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xiSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xkSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998 E.......*.Y16... .........^.SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986 E.......*.Y<6... ...........SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294 E..*....*.X.6... .........:.SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282 E.......*.X.6... ........ ..SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50 s=FreeSWITCH c=IN IP4 173.203.60.50 t=0 0 m=audio 25430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W.6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412 E.......@.o. ...6.........j.CANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6...........ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W~6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........jACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246 E.......*.\.6... ...........OPTIONS sip:68.34.22.11:1024 SIP/2.0 Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330 E..f....@.p. ...6........R.<SIP/2.0 405 Method not allowed Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Server: MS Lync Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W|6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........eACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W{6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........;ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200

1 0

how to use mqueue.
by 张顺通 06 Nov '14

06 Nov '14

whle (mq_fetch("QUEUE")) { if (yes) { do something } else { mq_add("QUEUE"); add back to queue. } } can mq_add use here? I tried it, it do not work. and How to determine the end?

3 6

Releasing v4.1.7
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, I plan to release a new version (4.1.7) of the previous stable branch 4.1. Most probably on Thursday this week. If anyone has remarks/issues about it, write back to mailing lists to discuss. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 1

benchmark module
by bob sacamano 05 Nov '14

05 Nov '14

hello, i was wondering where does this module write the logs? Nothing shows up in my syslog. If i replace the benchmark parts in the route with xlog for example xlog( "L_INFO", "$Ts " ); this shows up fine in syslog heres what u have in my config file: modparam("benchmark", "enable", 1) modparam("benchmark", "granularity", 100) modparam("benchmark", "loglevel", 3) if(is_method("REGISTER")) { bm_start_timer("test"); save("location"); bm_log_timer("test"); exit; } I have been searching but cant find any info on it

2 5

Kamailio with media proxy and two network interfaces
by Julien 05 Nov '14

05 Nov '14

Hello, I have a problem with Kamailio and a setup with two network interfaces (eth0 and eth1) Devices on eth0 CANNOT see devices on eth1 and vice versa, there is no route between both interfaces. Only Kamailio and the media proxy can access both. +--------------------------+ | kamailio + rtpproxy | eth0 | + openvpn | eth1 +---------+ +-----------+ | | | | | +--------------------------+ | +-+----+ +----+-+ | A | | B | | | | | +------+ +------+ A user connected at eth0 shall be able to get the presence status of users on eth1 and users shall be able to call each other. What I did so far: I added listen statements for both interfaces to the Kamailio config file: -listen=udp:eth0_IP:5060 -listen=udp:eth1_IP:5060 What's not working: When I try to make a call from user in A domain to another user in B domain, the call established correctly, but no stream running. On B device I see an incoming call from 'sip:user@A_Domain'. B user doesn't have route for A Domain, so that doesn't work. Is what I want technically possible with Kamailio. If yes, what do I have to change in my config? Do you have any suggestions? Thanks in advance. -- Julien Barreau

3 3

how you make Kamailio with mariadb
by ANTHONY HESNAUX 05 Nov '14

05 Nov '14

hello Daniel I want to install Kamailio with mariadb on CENTOS 7 I already install Kamailio on CENTOS 6.5 without error but on CENTOS 7 with mariadb, What modules do you include for make cfg: make include_modules="db_mysql" cfg thank you best regard Anthony HESNAUX

3 8

UAC error at reply : t_should_relay_response:
by Yuriy Gorlichenko 05 Nov '14

05 Nov '14

ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 Hello. I use UAC module for trunks form 4.2 master branch. I have an error when endpoit at the trunk picked up. OK reply never replied to caller because I see next error: ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 So as I see before reply 200 sended to kamailio (from callee) I see packet with 491 SIP code. It is strange because at the another server (same configuration) OK successfully sended to caller and session is Ok...

1 0

rare crash , race condition between 183 and 408 ?
by Dragos Oancea 05 Nov '14

05 Nov '14

Hi all & hi Daniel , We had a crash with version 4.0.5 today . I looks like it happens due to a race condition between a provisional response (183) and the generation of a 408 . We send the INVITE, get a `100 Trying` from upstream , then we get a `183 Session Progress` from upstream , then after exactly 1 minute , we get another `183 Session Progress` from upstream. At this moment kamailio crashed. Some tm params we have : modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) <- our final reply timer is 1 minute . modparam("tm", "restart_fr_on_each_reply", 1) GDB output and logs here: kamailio 4.0.5 crash - in fake_req() - race cond - Pastebin.com kamailio 4.0.5 crash - in fake_req() - race cond - Paste... (gdb) bt #0 0x00007f4d93129a32 in memcpy () from /lib64/libc.so.6 View on pastebin.com Preview by Yahoo In fake_req() from t_reply.c , the pointer shmem_msg->new_uri.s which is passed to memcpy becomes invalid. memcpy( faked_req->new_uri.s, shmem_msg->new_uri.s,faked_req->new_uri.len); (gdb) p shmem_msg->new_uri $3 = {s = 0x0, len = 0} If this bug is not supposed to be fixed in 4.2.0 , we can help with debugging . Regards, Dragos

2 1

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