sr-users February 2006

sr-users@lists.kamailio.org

189 participants
374 discussions

[Users] Clarity on the DISPATCHER module
by Anders Brownworth 08 Feb '06

08 Feb '06

Hi, I am trying to get an OpenSER instance running with the dispatcher module spreading load amongst several other OpenSER servers. I'm having trouble understanding from the docs and practical application exactly what the dispatcher module does and how to make it do what I want. I want to accept requests and use the dispatcher module to send requests to one of a number of other servers. (I want this setup to scale so I don't want to use the tm module.) Each subsequent request with the same CallId should go back to the same server. Therefore using the hash over the CallId algorithm in the dispatcher module seems like the best choice. I'm running OpenSER 1.0.0 and working off of the configuration in the modules documentation: http://openser.org/docs/modules/1.0.x/dispatcher.html I fixed up a few things that didn't seem to work out of the box such as replacing forward( ) with forward( uri:host, uri:port ). I have several questions. 1. What exactly does ds_select_dst("1", "0") do? I understand the parameters but the module documentation says it "selects a destination from addresses set". OK, so does it return it somehow? How do I get my hands on what it picks? What is an example of a situation where this function would be useful? 2. What exactly ds_select_domain("1", "0") do? I gather it rewrites the host and port with what it gets from the addresses set (which seems more useful than ds_select_dst), but when I run this in practice, _if_ it rewrites the host / port, it always seems to pick the same destination. (in my tests I only get an outgoing INVITE about 10% of the time) It does seem like I want to be using ds_select_domain() though. Can someone point out where my misunderstanding is? I have my configuration copied below. Thanks. -Anders --------------------------------------------------------------------------------------- What I'm expecting: When the first INVITE comes from 10.1.50.30 (Asterisk) to 10.1.50.31, (OpenSER with dispatcher) I want to see an outgoing INVITE from 10.1.50.31 to one of the listed addresses. (10.1.50.[34-37]) What I'm seeing: A tcpdump shows that most of the time all I see is the incoming INVITE. Seemingly random INVITES produce an outgoing INVITE, even when calling the same number. Presumably the only substantive difference is the CallID. openser.cfg: ----------------------------------------------------------------------------- #debug=4 fork=yes log_stderror=no children=4 check_via=no dns=off rev_dns=off port=5060 mpath="/usr/local/lib/openser/modules" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "dispatcher.so" modparam( "dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list" ) route { if ( ! mf_process_maxfwd_header( "10" ) ) { sl_send_reply( "483", "To Many Hops" ); drop( ); }; ds_select_domain( "1", "0" ); forward( uri:host, uri:port ); } ----------------------------------------------------------------------- dispatcher.list: ----------------------------------------------------------------------- # gateways 1 sip:10.1.50.34:5060 1 sip:10.1.50.35:5060 1 sip:10.1.50.36:5060 1 sip:10.1.50.37:5060 ----------------------------------------------------------------------- tcpdump shows: ----------------------------------------------------------------------- 07:43:37.216888 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 685 .@.@... .2. .2.........INVITE sip:+18666775910@10.1.50.31 SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK434da07b From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as3a379ead To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 7dd7978262a876832fc69a4364f89f22(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 202 v=0 o=root 11014 11014 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 8360 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- with no outgoing INVITE. Things seem to be dead. Then I try the same call a few seconds later and I get the outgoing INVITE: ----------------------------------------------------------------------- 07:43:48.695958 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 686 .@.@... .2. .2.........INVITE sip:+18666775910@10.1.50.31 SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as302ac772 To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 531db3695f15035620baae890d555216(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - 07:43:48.696191 IP 10.1.50.31.5060 > 10.1.50.36.5060: UDP, length: 747 E....*@.@..g .2. .2$......S.INVITE sip:+18666775910@10.1.50.36:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.50.31;branch=0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as302ac772 To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 531db3695f15035620baae890d555216(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- any help greatly appreciated.

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[Users] dispatcher forwarding troubles...
by Helge Waastad 08 Feb '06

08 Feb '06

Hi, I just wonder about one thing. I had my dispatcher proxy layer working before the last cvs update. Now I see in my logs: BUG: get_send_socket: unknown proto 0 Feb 4 16:50:04 proxy-01 /usr/sbin/openser[8192]: forward_req: ERROR: cannot forward to af 2, proto 0 no corresponding listening socket I just wonder what these messages mean. I'm forwarding messages to another OpenSER..... br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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Re: [Users] user is not found
by Bogdan-Andrei Iancu 08 Feb '06

08 Feb '06

Hi, have you subscribed the user into the DB? regards, bogdan Brijesh wrote: > Hi all > > I am new to OpenSER. > > I am using one of sip client. > > I want to connect sip clients through openSER proxy > > I have installed it and configured the clients but giving me error > that user is not found > > Can anyone tell me what is reason for it ? > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

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RE: [Users] Receiving different VIA in INVITE and CANCEL
by Papadopoulos Georgios 08 Feb '06

08 Feb '06

Hi Bogdan, Thank you for your reply. I used modparam("tm", "via1_matching", 0) modparam("tm", "ruri_matching", 0) and I got the following trace. You can see that the ACK to the Proxy Authentication Required and the CANCEL have different Via than the INVITE. I know that this breaks the specs but I thought that these two modparams should take care of these broken UAs. Also, any ideas what makes the UA switch ports? Is it a NAT issue? Only a very small percentage of the same UA behave this way. thank you for any help George # U 2006/02/07 12:05:29.756686 195.167.60.21:1231 -> 213.5.43.134:5060 INVITE sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 INVITE. Contact: tester <sip:tester@195.167.60.21:1327>. Remote-Party-ID: tester <sip:tester@sip.i-call.gr>;screen=yes;party=calling. Max-Forwards: 70. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.757360 213.5.43.134:5060 -> 195.167.60.21:1327 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f". Server: OpenSer (1.0.0-tls (x86_64/linux)). Content-Length: 0. Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10474 req_src_ip=195.167.60.21 req_src_port=1231 in_uri=sip:2116872933@sip.i-call.gr out_uri=sip:2116872933@sip.i-call.gr via_cnt==1". . # U 2006/02/07 12:05:29.865886 195.167.60.21:1327 -> 213.5.43.134:5060 ACK sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 ACK. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 70. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . ************************************************************************ ************** *** The Via in the ACK is different than in the INVITE and Openser sends it *** to the wrong server ************************************************************************ ************** # U 2006/02/07 12:05:29.866660 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 ACK sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 ACK. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 69. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:29.959041 195.167.60.21:1327 -> 213.5.43.134:5060 INVITE sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Contact: tester <sip:tester@195.167.60.21:1328>. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="da850791f85d3e882d384d96e4aff30e", algorithm=MD5. Remote-Party-ID: tester <sip:tester@sip.i-call.gr>;screen=yes;party=calling. Max-Forwards: 70. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.960454 213.5.43.134:5060 -> 195.167.60.21:1328 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Server: OpenSer (1.0.0-tls (x86_64/linux)). Content-Length: 0. Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10473 req_src_ip=195.167.60.21 req_src_port=1327 in_uri=sip:2116872933@sip.i-call.gr out_uri=sip:400#2116872933@213.5.yyy.yyy:5060 via_cnt==1". . ************************************************************************ *** Openser forwards the request to PSTN gateway ************************************************************************ # U 2006/02/07 12:05:29.960516 213.5.43.134:5060 -> 213.5.yyy.yyy:5060 INVITE sip:400#2116872933@213.5.yyy.yyy:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 69. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.975762 213.5.yyy.yyy:57918 -> 213.5.43.134:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2006/02/07 12:05:30.149177 213.5.yyy.yyy:57918 -> 213.5.43.134:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 288. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. s=SIP Call. c=IN IP4 213.5.yyy.yyy. t=0 0. m=audio 17868 RTP/AVP 18 101. c=IN IP4 213.5.yyy.yyy. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=silenceSupp:off - - - -. # U 2006/02/07 12:05:30.149263 213.5.43.134:5060 -> 195.167.60.21:1328 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 288. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. s=SIP Call. c=IN IP4 213.5.yyy.yyy. t=0 0. m=audio 17868 RTP/AVP 18 101. c=IN IP4 213.5.yyy.yyy. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=silenceSupp:off - - - -. ************************************************************************ **** *** User Agents sends CANCEL with different Via than in INVITE *** and Openser send it to the wrong server ************************************************************************ **** # U 2006/02/07 12:05:37.859125 195.167.60.21:1328 -> 213.5.43.134:5060 CANCEL sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. Max-Forwards: 70. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:37.859999 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 CANCEL sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. Max-Forwards: 69. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:37.860901 213.5.xxx.xxx:5060 -> 213.5.43.134:5060 SIP/2.0 481 Call Leg Does Not Exist. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0;received=213.5.43.134. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. User-Agent: i-Call Service. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Content-Length: 0. . # U 2006/02/07 12:05:37.860998 213.5.43.134:5060 -> 195.167.60.21:1375 SIP/2.0 481 Call Leg Does Not Exist. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. User-Agent: i-Call Service. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Content-Length: 0. . > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, February 03, 2006 10:43 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > > Hi George, > > different VIA in INVITE and CANCEL brakes the specs of RFC > 3261. The "via1_matching" is a kind of a trick to make > un-compliant clients to work. The trick is functional. I > suspect that maybe not only the VIA is the problem. Can you > please post the INVITE and CANCEL to have a look on them? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hello, > > > >I have an issue with some user agents, (Linksys PAP2 and > Sipura 3000) > >sending different Via in CANCEL than in INVITE. > >Of course the result is that the caller hangs up and the > callee keeps > >ringing. > >I tried to use > > modparam("tm", "via1_matching", 0) > >but it did not solve the problem. > > > >I have two questions: > >1. Is the via1_matching in TM working as described or is it broken? > > > >2. What can be causing this behavior of the user agent? Not all our > >users who have these user agents report this problem > (actually very few > >have the problem). In our lab we tried same user agent, same > firmware > >but could never reproduce this behavior no matter how bad we > tried to > >mess up their configuration. > > > >Any help will be greatly appreciated. Thanks. > > > >George > > > > > > > >Disclaimer > >The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] Presence changes in resource lists
by victoria beltran 08 Feb '06

08 Feb '06

Hi, I don't understand how the ser server manages the resource lists. When one contact of the resource list changes the state (via publish), how does the rls module notify this change to all subscribers of the list? Does the rls module implement the draft-ietf-simple-event-list-07 draft ("A SIP Event Notification Extension for Resource Lists")? Thanks!! --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

1 0

[Serusers] Problem with SER and Database connections
by mkumar＠mantragroup.com 08 Feb '06

08 Feb '06

Hi All, I installed SER 0.9.4 and I was able to make and receive calls. SER and Mysql run on the same unix box which is located at some other place and which also runs a webserver. After starting ser everything goes fine but after sometime the whole system goes down and I was not even able to connect to unix box using SSH or to any web page of that webserver. Does the Number of connections made by SER to Mysql is causing the problem or is there anything wrong with my SER configuration and Mysql? If Number of connections is the problem then please tell me what must I do to reduce the number of connections to Mysql from SER. Also please tell me if there is any new Mysql.so module or any other module which fixes this. Thanks, Manoj

3 6

[Serusers] scripts needed to gathering all SIP messages
by hgaillac-sip＠yahoo.fr 08 Feb '06

08 Feb '06

Hello, I have txt file with calls flow. I look for a script to Create ,for example , a 679400FC-8C211DA-9C3DBF26-F375EEC9@192.168.52.52 file, gathering all SIP messages (find the way to easily detect the start and the end of every SIP messages). Thanks for help Harry 2005/08/10-12:44:44.324 ===== Received ======================= 12:44:44 324 id1/192.168.49.1:5061 <===UDP=== /192.168.49.1:5060 SIP/2.0 200 OK To: <sip:0097143389080@192.168.52.52>;tag=A8C32828-2670 Allow-Events: telephone-event Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Server: Cisco-SIPGateway/IOS-12.x Call-ID: 679400FC-8C211DA-9C3DBF26-F375EEC9(a)192.168.52.52 Contact: <sip:0097143389080@192.168.52.52:5060> Content-Type: application/sdp CSeq: 107 INVITE Date: Wed, 10 Aug 2005 10:44:44 GMT Content-Length: 234 Via: SIP/2.0/UDP 192.168.49.1:5061;branch=z9hG4bK228 Record-Route: <sip:192.168.49.1;lr> From: <sip:497231265@id.centile.com>;tag=c74716fe-957a-816c-1304-aa9de235bd46 ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

3 2

[Serusers] SEMS G729
by Kapil Dhawan 08 Feb '06

08 Feb '06

Hi List How can i compile SEMS with g729. _________________________________________________________________ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

6 9

[Serusers] ser installation
by Alex Wu 08 Feb '06

08 Feb '06

Hi I'm fairly new to sip express router server so the question may have been asked before. I was used debian os,.when installation ser deb package by apt-get install ser .I find this service not really start. Info follow: *********command lines*********************** Alex:/etc/ser# /etc/init.d/ser restart Restarting ser: serListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.10.51 [192.168.10.51]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.10.51 [192.168.10.51]:5060 Aliases: tcp: Alex:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: Alex:5060 udp: localhost:5060 udp: localhost.localdomain:5060 ****syslog************************** Feb 7 10:19:55 www /usr/sbin/ser[14827]: WARNING: destroy_fifo: cannot delete fifo (/tmp/ser_fifo): Operation not permitted (?????????????????????????????????) Feb 7 10:19:56 www /usr/sbin/ser[14865]: Maxfwd module- initializing Feb 7 10:19:56 www /usr/sbin/ser[14866]: WARNING: no fifo_db_url given - fifo DB commands disabled! (?????????????????????????????) ****/tmp/ser_fifo attribute************* Alex:/tmp# ls -l total 0 prw-rw---- 1 root root 0 2006-02-07 10:21 ser_fifo prw-rw-rw- 1 root root 0 2006-02-07 10:21 ser_receiver_14893

1 0

[Serusers] [Help] wm not signing in
by ryan nalupa 08 Feb '06

08 Feb '06

hi all! i'm still new to ser scripting and ser itself. i just followed what i read in the admin guide and some samples that i obtained from the net. my ser box is using version 0.9.6. my problem is i can't make my wm sign in, i just followed what was said in some docs that i've read but still it won't sign in but when i'm using an xlite softphone it's fine i mean the softphone can log in and can make calls. do i still have to do some more module loading or some parameter settings? please help, i'm still new here! thanks in advance! :-) ryan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

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sr-users February 2006