sr-users

sr-users@lists.kamailio.org

44 participants
37550 discussions

New developer: Luis Azedo
by Daniel-Constantin Mierla 10 Sep '14

10 Sep '14

Hello, I would like to announce that a new developer has now write access to Kamailo git repository: Luis Azedo. His commit username is: lazedo He is going to push soon the 'kazoo' module that was discussed these days on the mailing list - a module connector for the Kazoo - the open source, distributed VoIP platform using Kamailio for core SIP routing. He also provided many patches to other components so far. Luis works for 2600hz, an USA company, and will be accompanied by Darren Schreiber and Karl Andreson in maintenance of the new module. My warm welcome and thanks for joining the effort of developing Kamailio further! Looking forward to future contributions to the project! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany

1 0

No BYE from Called party
by Yuriy Gorlichenko 09 Sep '14

09 Sep '14

Hello All. I have kamailio with provider connection (trunk) When I call to external number through my provider call extablished Ok. But when i try hangup call from external number no BYE sended to me. When I hangup call from my kamailio (internal num) I send by to exteral number and it respond me Ok so session if fully complete. I guess that BYE from external number not recieves to me because I have wrong routing header fields at my INVITe or ACK messages, but can not find any information what what header must recieve info to external number where send BYE at hangup or thomething like this. This is my little dump for situation wherer I hangup from internal number and BYE finished successfully: IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 1599 E....< .@.'. ...6........G.RINVITE sip:12345678900@my.provider.com:5060 SIP/2.0 Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Max-Forwards: 70 From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068> Contact:<TrunkNum@my.kamailio.com:5068> Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 04 Sep 2014 21:53:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 544 Proxy-Authorization: Digest username="TrunkNum", realm="my.provider.com", nonce="VAjgfVQI31EsekTXtYjBzKa59XFSJB5P", uri=" sip:12345678900@my.provider.com:5060", qop=auth, nc=00000001, cnonce="3619116795", response="f5bc1d8125dd9e448d2e73764823adee", algorithm=MD5 v=0 o=root 1022912010 1022912010 IN IP4 my.kamailio.com s=Asterisk PBX 12.5.0 c=IN IP4 my.kamailio.com t=0 0 a=ice-lite m=audio 30032 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 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=rtcp:30033 a=ice-ufrag:3o8JrqkF a=ice-pwd:m8q4khQT8NKSABqeUkKs7ed8ClR2 a=candidate:TgT1dfTnI3kBgWQ IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 1882 E..... ./...6... ........b..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;rport=5068;received=my.kamailio.com ;branch=z9hG4bK5c63.57e9ba848fadd4e228caa4445baf9d76.2 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK4d90cebf;rport=50600 Record-Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Record-Route: <sip:my.kamailio.com:5068;nat=yes;ftag=as5872f19e;lr=on> From: <sip:TrunkNum@my.provider.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 INVITE Contact: <sip:12345678900@externail.number.end.ip:5060;transport=udp> User-Agent: provider agent 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: 746 X-provider agentOutboundGateway: sip:5574012345678900@62.93.147.149 X-provider agentOutboundCarrierID: 23705946361020 X-provider agentCarrierRate: 0.20180 X-provider agentCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@my.provider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1409844386 1409844387 IN IP4 externail.number.end.ip s=FreeSWITCH c=IN IP4 externail.number.end.ip t=0 0 a=msid-semantic: WMS hQBNQPw0VjJInvOA0H6HPZyePNO0IIqP m=audio 23216 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 a=ssrc:2990874569 cname:pRs5xP IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 596 E..p.@..@.K\ ...6........\`.ACK sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK5c63.8c0bfc7de0669f1b8252b97b8431b2e1.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK5420b559;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 102 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP my.kamailio.com.5068 > my.proider.com.5060: UDP, length 617 E....D..@.KC ...6........q.ZBYE sip:12345678900@externail.number.end.ip:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP my.kamailio.com:5068 ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 Route: <sip:my.proider.com;lr=on;ftag=as5872f19e> Max-Forwards: 70 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: Asterisk PBX 12.5.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 IP my.proider.com.5060 > my.kamailio.com.5068: UDP, length 568 E..T....-.676... ........@..SIP/2.0 200 OK Via: SIP/2.0/UDP my.kamailio.com:5068;received=my.kamailio.com ;branch=z9hG4bK6c63.3deda3c1fee0ef86d7eae9549dd1b194.0 Via: SIP/2.0/UDP my.client.internal.ip:50600;branch=z9hG4bK65e48b85;rport=50600 From: <sip:TrunkNum@sip.callsion.com>;tag=as5872f19e To: <sip:12345678900@my.provider.com:5068>;tag=5rF0FNamQ99gH Call-ID: 42b819d45c08c9d304343bf976c5b405@my.client.internal.ip:50600 CSeq: 103 BYE User-Agent: provider agent Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 Thanks for help.

2 4

Using app_java
by Tom Johnson 09 Sep '14

09 Sep '14

Any tips on compiling the app_java module? I am on CentOS 6. And I have libgcj-4.4.7 installed, but the build process complains about missing jni.h and other files.

2 1

Creating array out of hashtable
by aft 09 Sep '14

09 Sep '14

[1]Is it possible to iterate a htable without using "keys" like an array? [2] Is it possible to search for "keys" using regxp? -- -Cheers -Arif

2 2

Call connection from one server to another
by aawaise 09 Sep '14

09 Sep '14

Hello, I want to deploy two kamailio servers, both using same MYSQL database placed on Server 2008. Both are using different static IP's. Hopefully, I can add two A records in DNS server, both against same domain name, pointing to two different IP's to which two servers are attached. My question is that in load balancing scenario if user 1 is attached to server 1, and user 2 is attached to server 2. How can user1 call user2. Because request will go to server 1 to which user 1 is attached. But it wont be able to locate user 2 because it is not connected to server1 (but connected to server 2). I am using same domain name for two servers. What can be done in kamailio to solve this issue ?? Regards, aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Call-connection-from-one-server-to-… Sent from the Users mailing list archive at Nabble.com.

3 3

TLS Handshake failing with WSS
by Manuel Camarg 08 Sep '14

08 Sep '14

I'm trying to implement WSS with Kamailio Thing is that WS works fine, I've followed: http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket modparam("tls", "config", "webrtc/tls.cfg") In a tls.cfg file I have : [server:default] method = SSLv23 verify_certificate = no require_certificate = no private_key = webrtc/private.key certificate = webrtc/ssl.pem ca_list = webrtc/ca_list.pem In the log file: /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=5524 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8e0040, 11, 2, 0x7f7513516958), fd_no=1 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:712]: sr_ssl_ctx_info_callback(): SSL handshake done /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL disable renegotiation /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 123.123.123.123:63300 using TLSv1/SSLv3 AES256-SHA 256 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 124.124.124.124:10443 /usr/local/sbin/kamailio[4025]: DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f7513516958 n=282 fd=11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf=#012#026#003#003 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7f7513516958, FD 11 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1293]: tcp_read_req(): tcp_read_req: EOF /usr/local/sbin/kamailio[4025]: DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e0040, 11, -1, 0x10) fd_no=2 called /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f7513516958, state -1, fd=11, id=2 /usr/local/sbin/kamailio[4025]: DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f7513510a88 /usr/local/sbin/kamailio[4029]: DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f7513516958, -1 from 1 /usr/local/sbin/kamailio[4029]: DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7f7513510a88 In sipml5 the error: *Disconnected: Failed to connect to the server* In the Chrome console: *__tsip_transport_ws_onerror * *__tsip_transport_ws_onclose * SSL certificates seem to be ok: # openssl verify -CAfile ca_list.pem ssl.pem ssl.pem: OK Can't figure out a solution :( Any ideas? *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

2 3

Change caller name in kamailio as the extension remains same.
by Talha Bukhari 08 Sep '14

08 Sep '14

Hi, I am new to kamailio. i configure kmailio server and make call as local extensions like 1001 and etc. Now i want to know how to change the header name appear on softphone using script in kamailo as the extension remain same as 1001. what are the files i used to alter and where are to locate them. Hope you don't mind. Regards, Talha

2 4

out of memory ; needs 455
by Arun Kumar 08 Sep '14

08 Sep '14

Hi Server : kamailio 4.1.1 Configuration :Outbound with engage_media_proxy (See the attachment ) Shared Mem : 1024 MB Problem : after 1 month kamailio hang up with out of memory error but we have enough free shared memory , it might be memory pool size problem, i dont know how and where to set memory pool size

2 1

Kamailio - voicemail platform - asterisk
by Henny 08 Sep '14

08 Sep '14

Hello, I am new to kamailio...... I have an idea for a community voice-mail platform (with Asterisk)and was wondering if Kamailio could be of any use in this plan. Does somebody have any advice ? Thanx!

2 1

dialog match on loop back
by Kelvin Chua 06 Sep '14

06 Sep '14

hi guys, i'm running out of ideas i have a situation where I need to generate 2 sets of entries INVITE,BYE in acc (acc_db_request won't cut it) it's when a DID should be redirected to a user outside my domain, I need to generate a cdr for DID and another CDR for the outbound call. what i am trying to do is loop it back to itself before routing it out. the idea is kamailio assigns the DID call to a dialog. then the outbound call will be assigned to a separate dialog after it receives the loopback. this is not happening since the from_tag, to_tag and the callid are all the same. kamailio treats this call as just 1, not 2. is there any ideas out there to work around this scenario? Kelvin Chua

2 1

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users