sr-users July 2013

sr-users@lists.kamailio.org

92 participants
148 discussions

Problem with forward on busy
by LAA 25 Jul '13

25 Jul '13

Hi Carsten, I forgot exit!! Anyway, aas this was the last part of failure route it don't make any differente. Many Thanks. L.

2 1

Re: [SR-Users] Problem with forward on busy
by LAA 25 Jul '13

25 Jul '13

Hello Hero, Thanks for your help. May be I'm loosing something. I have changed my config as you suggested (I thing so...): if (t_check_status("486|408")) { revert_uri(); prefix("voicemail"); remove_hf("P-App-Name"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com;uid=$ rU;did=sipproxy.a.com;\r\n"); rewritehostport("192.168.0.197:5080"); $du = $null; #$du = "sip:192.168.0.197"; append_branch(); t_relay(); } } Kamailio sends back 200 OK to the UAC that originated the call, but it never sends the new INVITE |Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | |3,151 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,159 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |3,161 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,161 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,174 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |3,174 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |3,176 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |3,177 | | 486 Busy Here |SIP Status | | |(5060) <------------------ (5060) | |3,180 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |3,195 | 200 OK SDP ( telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |3,200 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,213 | RTP (GSM) | | |RTP Num packets:204 Duration:4.069s SSRC:0x8494958 | |(49222) ------------------> (10028) | | |7,288 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |7,295 | 200 OK | | |SIP Status | |(5060) <------------------ (5060) | | what am I loosing? Regards LAA ************* had the same issue here. you have to manually set $du=$null, else it doesn't get reset for the failure branch. On 7/23/13, LAA <ornitorrinco7424 at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >* Hi all,*>**>* I'm running Kamailio 3.0.0, with SEMS integration as Media Server for Voice*>* mail. I'm trying to get a configuration to forward calls on busy to voice*>* mail. I have followed without success some examples. I'm using*>* revert_uri(), rewritehostport() and append_branch(), within failure_route.*>* It seems to be modifying R-URI properly, and generating the new branch, but*>* Kamailio is sending the new invite packet to the IP address of the original*>* destination UAC, and not to the IP address of the voicemail, that was*>* indicated in the R-URI. Here you can see the packet flow:*>**>* |Time | 192.168.3.20*>* | 192.168.0.167 |*>* | | | 192.168.0.197 |*>* |5,069 | INVITE SDP ( telephone-event)*>* | |SIP From: sip:4095 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* To:sip:4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* | |(5060) ------------------> (5060) | |*>* |5,071 | 407 Proxy Authentication Required*>* | |SIP Status*>* | |(5060) <------------------ (5060) | |*>* |5,074 | ACK | | |SIP*>* Request*>* | |(5060) ------------------> (5060) | |*>* |5,076 | INVITE SDP ( telephone-event)*>* | |SIP From: sip:4095 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* To:sip:4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* | |(5060) ------------------> (5060) | |*>* |5,084 | 100 trying -- your call is important to us*>* | |SIP Status*>* | |(5060) <------------------ (5060) | |*>* |5,085 | | INVITE SDP (*>* telephone-event) |SIP Request*>* | | |(5060) ------------------> (5060) |*>* |5,088 | | 100 Trying| |SIP*>* Status*>* | | |(5060) <------------------ (5060) |*>* |5,088 | | 486 Busy Here |SIP*>* Status*>* | | |(5060) <------------------ (5060) |*>* |5,091 | | ACK | |SIP*>* Request*>* | | |(5060) ------------------> (5060) |*>* |5,101 | | INVITE SDP (*>* telephone-event) |SIP Request*>* | | |(5060) ------------------> (5060) |*>* |5,102 | | 404 Not Found |SIP*>* Status*>* | | |(5060) <------------------ (5060) |*>* |5,102 | | ACK | |SIP*>* Request*>* | | |(5060) ------------------> (5060) |*>* |5,103 | 404 Not Found | |SIP*>* Status*>* | |(5060) <------------------ (5060) | |*>* |5,106 | ACK | | |SIP*>* Request*>* | |(5060) ------------------> (5060) | |*>**>* And the RAW capture of the INVITE message in timestamp 5,101.*>**>**>**>* No. Time Source Destination Protocol*>* Info*>* 1235 5.100698 192.168.0.197 192.168.0.167 SIP/SDP*>* Request: INVITE sip:voicemail4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080, with session*>* description*>**>* Frame 1235 (1151 bytes on wire, 1151 bytes captured)*>* Ethernet II, Src: CadmusCo_96:31:84 (08:00:27:96:31:84), Dst:*>* Micro-St_6d:77:54 (00:21:85:6d:77:54)*>* Internet Protocol, Src: 192.168.0.197 (192.168.0.197), Dst: 192.168.0.167*>* (192.168.0.167)*>* User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)*>* Session Initiation Protocol*>* Request-Line: INVITE sip:voicemail4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080 SIP/2.0*>* Method: INVITE*>* Request-URI: sip:voicemail4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5080*>* [Resent Packet: True]*>* [Suspected resend of frame: 1233]*>* Message Header*>* Record-Route: <sip:192.168.0.197;lr=on;nat=*>* yes>*>* Via: SIP/2.0/UDP 192.168.0.197;branch=z9hG4bKafce.403718a6.1*>* Via: SIP/2.0/UDP*>* 192.168.57.20;received=192.168.3.20;rport=5060;branch=z9hG4bK0a00030f0000003151ed60b85ec2c3de000000c8*>* Content-Length: 386*>* Contact: <sip:4095 at 192.168.3.20 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>*>* Call-ID: 8EAF9EC2-1DD2-11B2-B110-C84E476664B0 at 10.0.3.15 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* Content-Type: application/sdp*>* CSeq: 2 INVITE*>* From: "4095"<sip:4095 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=121754238352072516*>* Max-Forwards: 69*>* To: <sip:4440 at 192.168.0.197 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>*>* User-Agent: SJphone/1.60.299a/L (SJ Labs)*>* P-App-Name: voicemail*>* P-App-Param: mod=box;usr= voicemail4440;dom=sipproxy.a.com*>* ;uid=voicemail4440;did=sipproxy.a.com;*>* Message Body*>**>* Here you can see the failure_route in my kamailio.cfg file:*>**>* # Sample failure route*>* failure_route[FAIL_ONE] {*>* #ifdef WITH_NAT*>* if (is_method("INVITE")*>* && (isbflagset("6") || isflagset(5))) {*>* unforce_rtp_proxy();*>* }*>* #endif*>**>* if (t_is_canceled()) {*>* exit;*>* }*>**>* # uncomment the following lines if you want to block client*>* # redirect based on 3xx replies.*>* ##if (t_check_status("3[0-9][0-9]")*>* ) {*>* ##t_reply("404","Not found");*>* ## exit;*>* ##}*>**>* # uncomment the following lines if you want to redirect the failed*>* # calls to a different new destination*>* if (t_check_status("486|408")) {*>* revert_uri();*>* prefix("voicemail");*>* remove_hf("P-App-Name");*>* append_hf("P-App-Name: voicemail\r\n");*>* append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com*>* ;uid=$rU;did=sipproxy.a.com;\r\n");*>* $ru = "sip:" + $rU + "@" + "192.168.0.197:5080";*>* #rewritehostport("192.168.0.197:5080");*>* #append_branch("sip:4888 at 192.168.0.102 <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>");*>* append_branch();*>* # do not set the missed call flag again*>* t_relay();*>* }*>* }*>**>* Has anybody experienced this problem? Any help would be wellcome*>**>* Best Regards*>**>* LAA*>

2 5

Problem with forward on busy
by LAA 25 Jul '13

25 Jul '13

Excuse me. I have created a new thread by mistake. ... Hello Hero, Thanks for your help. May be I'm loosing something. I have changed my config as you suggested (I thing so...): if (t_check_status("486|408")) { revert_uri(); prefix("voicemail"); remove_hf("P-App-Name"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com;uid=$ rU;did=sipproxy.a.com;\r\n"); rewritehostport("192.168.0.197:5080"); $du = $null; #$du = "sip:192.168.0.197"; append_branch(); t_relay(); } } Kamailio sends back 200 OK to the UAC that originated the call, but it never sends the new INVITE |Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | |3,151 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,159 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |3,161 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,161 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |3,174 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |3,174 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |3,176 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |3,177 | | 486 Busy Here |SIP Status | | |(5060) <------------------ (5060) | |3,180 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |3,195 | 200 OK SDP ( telephone-event) | |SIP Status | |(5060) <------------------ (5060) | | |3,200 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |3,213 | RTP (GSM) | | |RTP Num packets:204 Duration:4.069s SSRC:0x8494958 | |(49222) ------------------> (10028) | | |7,288 | BYE | | |SIP Request | |(5060) ------------------> (5060) | | |7,295 | 200 OK | | |SIP Status | |(5060) <------------------ (5060) | | what am I loosing? Regards

2 1

Re: [SR-Users] Regd. Register Request
by Daniel-Constantin Mierla 25 Jul '13

25 Jul '13

Hello, have you created the transaction before calling registrar/usrloc functions (e.g., using t_newtran())? You seem to want to match from transaction point of view. Registrar is matching based on contact, that being the key for the registrations -- there are some module parameters that allow different matching options. Cheers, Daniel On 7/25/13 11:46 AM, Shankar wrote: > > Hi, > > We are testing the REGISTRAR with kamailio. > > We tested the following scenario , > > èRegister Request with contact 'A'. > > è200 OK received for the request > > èSecond Register request with contact 'B'. (Requet-URI, To Tag, From > tag, call-id, and Cseq -> all same as the first register request) > > è200 OK received for the request. > > When we checked the 'location' table entry, two records were found for > the same user with different contact addresses. > > Please find below the RFC extract below, > > Section 17.2.3 > > For all other request methods, a request is matched to a transaction > > if the Request-URI, To tag, From tag, Call-ID, CSeq (including the > > method), and top Via header field match those of the request that > > created the transaction. Matching is done based on the matching > > rules defined for each of those header fields. When a non-INVITE > > request matches an existing transaction, it is a retransmission of > > the request that created that transaction. > > My understanding is that the second register request should be > considered as duplicate and rejected. > > Please share your feedback on the same. > > Regards, > > Shankar > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Regd. Register Request
by Shankar 25 Jul '13

25 Jul '13

Hi, We are testing the REGISTRAR with kamailio. We tested the following scenario , è Register Request with contact A. è 200 OK received for the request è Second Register request with contact B. (Requet-URI, To Tag, From tag, call-id, and Cseq -> all same as the first register request) è 200 OK received for the request. When we checked the location table entry, two records were found for the same user with different contact addresses. Please find below the RFC extract below, Section 17.2.3 For all other request methods, a request is matched to a transaction if the Request-URI, To tag, From tag, Call-ID, CSeq (including the method), and top Via header field match those of the request that created the transaction. Matching is done based on the matching rules defined for each of those header fields. When a non-INVITE request matches an existing transaction, it is a retransmission of the request that created that transaction. My understanding is that the second register request should be considered as duplicate and rejected. Please share your feedback on the same. Regards, Shankar

1 0

XCAP table is not getting updated
by Premchandiran 25 Jul '13

25 Jul '13

Hi All, I have cfg file which behaves like both presence and xcap (Integarted cfg) took reference from http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-support-im-and -presence-over-kamailio-31-debian-lenny. Kamailio is running but xcap table is not getting updated. Using JTSI as client(please refer screen shot for xcap configuration in jtsi client). I would like to know the following 1.) Any issue/missing in the cfg file (Please find the attached cfg file). 2.) When xcap table will be updated. I kindly request to help us. Regards, Prem Chandiran M

1 0

How to tie Registration with Client TCP connection
by Krishna Kurapati 24 Jul '13

24 Jul '13

Hi, With TCP, there is no way Server can establish a TCP connection through client's Firewall/NAT. So, when Client connection is disconnected from the server due to Network issues, the Registration should be removed. I tried to handle this with a Event Route in tcp_read.c. However, There is not an easy way to find the Registration with TCP socket info (IP and Port of the Client). Could anyone suggest a solution? Thanks Krish Kura

3 2

Kamailio 4.0.1 presence support
by Rupayan Dutta 24 Jul '13

24 Jul '13

Hello, I've been trying to make the Presence thing work with kamailio 4.0.1 and SipML5 browser client .I have loaded only the modules which are already in the Kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" Is it necessary to load other modules as well?At the same time I request you to suggest which Sip client I should use in order to make presence work. Rupayan Dutta

2 4

mediaproxy-ng documentation available
by Andreas Granig 23 Jul '13

23 Jul '13

Hi, There've been a couple of questions regarding mediaproxy-ng as an rtpproxy replacement on how to get it up and running with kamailio and what it does compared to rtpproxy, mediaproxy and iptrtpproxy. I hope, the README.md at https://github.com/sipwise/mediaproxy-ng answers these questions and makes it easy for you to use it. Feedback highly appreciated, Andreas

9 40

Problem with forward on busy
by LAA 23 Jul '13

23 Jul '13

Hi all, I'm running Kamailio 3.0.0, with SEMS integration as Media Server for Voice mail. I'm trying to get a configuration to forward calls on busy to voice mail. I have followed without success some examples. I'm using revert_uri(), rewritehostport() and append_branch(), within failure_route. It seems to be modifying R-URI properly, and generating the new branch, but Kamailio is sending the new invite packet to the IP address of the original destination UAC, and not to the IP address of the voicemail, that was indicated in the R-URI. Here you can see the packet flow: |Time | 192.168.3.20 | 192.168.0.167 | | | | 192.168.0.197 | |5,069 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |5,071 | 407 Proxy Authentication Required | |SIP Status | |(5060) <------------------ (5060) | | |5,074 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | |5,076 | INVITE SDP ( telephone-event) | |SIP From: sip:4095@192.168.0.197 To:sip:4440@192.168.0.197 | |(5060) ------------------> (5060) | | |5,084 | 100 trying -- your call is important to us | |SIP Status | |(5060) <------------------ (5060) | | |5,085 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |5,088 | | 100 Trying| |SIP Status | | |(5060) <------------------ (5060) | |5,088 | | 486 Busy Here |SIP Status | | |(5060) <------------------ (5060) | |5,091 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |5,101 | | INVITE SDP ( telephone-event) |SIP Request | | |(5060) ------------------> (5060) | |5,102 | | 404 Not Found |SIP Status | | |(5060) <------------------ (5060) | |5,102 | | ACK | |SIP Request | | |(5060) ------------------> (5060) | |5,103 | 404 Not Found | |SIP Status | |(5060) <------------------ (5060) | | |5,106 | ACK | | |SIP Request | |(5060) ------------------> (5060) | | And the RAW capture of the INVITE message in timestamp 5,101. No. Time Source Destination Protocol Info 1235 5.100698 192.168.0.197 192.168.0.167 SIP/SDP Request: INVITE sip:voicemail4440@192.168.0.197:5080, with session description Frame 1235 (1151 bytes on wire, 1151 bytes captured) Ethernet II, Src: CadmusCo_96:31:84 (08:00:27:96:31:84), Dst: Micro-St_6d:77:54 (00:21:85:6d:77:54) Internet Protocol, Src: 192.168.0.197 (192.168.0.197), Dst: 192.168.0.167 (192.168.0.167) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:voicemail4440@192.168.0.197:5080 SIP/2.0 Method: INVITE Request-URI: sip:voicemail4440@192.168.0.197:5080 [Resent Packet: True] [Suspected resend of frame: 1233] Message Header Record-Route: <sip:192.168.0.197;lr=on;nat= yes> Via: SIP/2.0/UDP 192.168.0.197;branch=z9hG4bKafce.403718a6.1 Via: SIP/2.0/UDP 192.168.57.20;received=192.168.3.20;rport=5060;branch=z9hG4bK0a00030f0000003151ed60b85ec2c3de000000c8 Content-Length: 386 Contact: <sip:4095@192.168.3.20:5060> Call-ID: 8EAF9EC2-1DD2-11B2-B110-C84E476664B0(a)10.0.3.15 Content-Type: application/sdp CSeq: 2 INVITE From: "4095"<sip:4095@192.168.0.197>;tag=121754238352072516 Max-Forwards: 69 To: <sip:4440@192.168.0.197> User-Agent: SJphone/1.60.299a/L (SJ Labs) P-App-Name: voicemail P-App-Param: mod=box;usr= voicemail4440;dom=sipproxy.a.com ;uid=voicemail4440;did=sipproxy.a.com; Message Body Here you can see the failure_route in my kamailio.cfg file: # Sample failure route failure_route[FAIL_ONE] { #ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #endif if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]") ) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination if (t_check_status("486|408")) { revert_uri(); prefix("voicemail"); remove_hf("P-App-Name"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: mod=box;usr= $rU;dom=sipproxy.a.com ;uid=$rU;did=sipproxy.a.com;\r\n"); $ru = "sip:" + $rU + "@" + "192.168.0.197:5080"; #rewritehostport("192.168.0.197:5080"); #append_branch("sip:4888@192.168.0.102"); append_branch(); # do not set the missed call flag again t_relay(); } } Has anybody experienced this problem? Any help would be wellcome Best Regards LAA

2 1

← Newer
1
...
4
5
6
7
8
9
10
...
15
Older →

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 July 2013