Hi, ?Please check the sip format to dialing the users.or can you plesae send us the sip?Registrar format to?me. a. have u registrar the users in Asterisk server along iwth the ip. check the invite request in the "Asterisk server"... and send?me ?the log. Have a look @ my small web-page: http://www.geocities.com/muki_champs Regards, Mukesh Kumar, Sr.Software Engineer, Mobile Application Developer. Hyderabad. India. +91-9397845485 (M)? ----- Original Message ---- From: ?? <skysoshy@xxxxxxx> To: pjsip at lists.pjsip.org Cc: liuzhidong at opencon3322.org Sent: Friday, June 27, 2008 11:56:10 AM Subject: Help: PjSip INVITE Message problem ? Hi all, ? I got a problem in my project. There is a pjsip 0.7.0 that used in it. ? I set up a AsteriskNow 1.0.2 as Sip Proxy Server. And install?two?SjPhones,One?on my PC,the other one on another PC. ? I do some simple configuration on Asterisk Sever: Add four accout for two?Pjsip phone and my SjPhones. Like this: [5001] context=occ_sip type=friend host=dynamic username=5001 secret=sparc10 nat=no canreinvite=yes insecure=port,invite Add four?dial rule: [occ_sip] exten => 66660000,1,dial(SIP/66660000) exten => 66661111,1,dial(SIP/66661111) exten => 5001,1,dial(SIP/5001) exten => 5002,1,dial(SIP/5002)? ? ? Any body has a clue or suggestion.Please?let me know.Thanks! ? Following are problem description: ? 1. When SjPhone call Pjsip Phone. Pjsip will response a 500 "Internal Server Error" for INVITE Message. If you call again.There will encounter a "Segmentation fault" Following is debug log of Pjsip: ? ###################################################################################################### First Call: ###################################################################################################### 19:03:06.346?? pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102 (rdata0x104793dc) from UDP 192.168.1.11:5060: INVITE sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK6c1d9bb1;rport From: "Michael Liu" <sip:66660000@192.168.1.11>;tag=as62628929 To: <sip:5001 at 192.168.1.200:5060;transport=UDP> Contact: <sip:66660000 at 192.168.1.11> Call-ID: 1797e7c14ead3bed679382095579af28 at 192.168.1.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 27 Jun 2008 05:42:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 4236 4236 IN IP4 192.168.1.11 s=session c=IN IP4 192.168.1.11 t=0 0 m=audio 10786 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv --end msg-- ?19:03:06.347?? pjsua_core.c TX 317 bytes Response msg 500/INVITE/cseq=102 (tdta0x1047eb64) to UDP 192.168.1.11:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.1.11:5060;rport=5060;received=192.168.1.11;branch=z9hG4bK6c1d9bb1 Call-ID: 1797e7c14ead3bed679382095579af28 at 192.168.1.11 From: "Michael Liu" <sip:66660000@192.168.1.11>;tag=as62628929 To: <sip:5001 at 192.168.1.200> CSeq: 102 INVITE Content-Length:? 0 --end msg-- ?19:03:06.349?? pjsua_core.c RX 407 bytes Request msg ACK/cseq=102 (rdata0x104793dc) from UDP 192.168.1.11:5060: ACK sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK6c1d9bb1;rport From: "Michael Liu" <sip:66660000@192.168.1.11>;tag=as62628929 To: <sip:5001 at 192.168.1.200:5060;transport=UDP> Contact: <sip:66660000 at 192.168.1.11> Call-ID: 1797e7c14ead3bed679382095579af28 at 192.168.1.11 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --end msg-- ?19:03:06.349 sip_endpoint.c Message Request msg ACK/cseq=102 (rdata0x104793dc) from 192.168.1.11:5060 was dropped/unhandled by any modules ? ###################################################################################################### Second Call: ###################################################################################################### ?19:20:22.633?? pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102 (rdata0x104793dc) from UDP 192.168.1.11:5060: INVITE sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK6feef7f9;rport From: "Michael Liu" <sip:66660000@192.168.1.11>;tag=as54c98a19 To: <sip:5001 at 192.168.1.200:5060;transport=UDP> Contact: <sip:66660000 at 192.168.1.11> Call-ID: 3b15421d5405cf87366858b0007f618e at 192.168.1.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 27 Jun 2008 05:59:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 4236 4236 IN IP4 192.168.1.11 s=session c=IN IP4 192.168.1.11 t=0 0 m=audio 16936 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv --end msg-- Segmentation fault ? ###################################################################################################### ? 2. When Pjsip Phone call SjPhone.Asterisk will prompt some error: ###################################################################################################### [Jun 27 01:22:02] NOTICE[4271]: chan_sip.c:13885 handle_request_invite: Call from '5001' to extension '192.168.1.11:5060' rejected because extension not found. AsteriskServer*CLI> ###################################################################################################### ? ? Pjsip receive a 404 "Not found" Message. ###################################################################################################### ?19:42:41.370?? pjsua_core.c TX 1059 bytes Request msg INVITE/cseq=846930887 (tdta0x1047e614) to UDP 192.168.1.11:5060: INVITE sip:192.168.1.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee0000001f6b8b4567 Max-Forwards: 70 From: "Freescale1" <sip:5001@192.168.1.11>;tag=00ee0000001c6b8b4567 To: sip:5002 at 192.168.1.11 Contact: "Freescale1" <sip:5001 at 192.168.1.200:5060;transport=UDP> Call-ID: 00ee0000001d6b8b4567 CSeq: 846930887 INVITE Route: <sip:5002 at 192.168.1.11:5060> Route: <sip:192.168.1.11:5060> Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONS Supported: norefersub User-Agent: PJSUA v0.7.0/linux Proxy-Authorization: Digest username="5001", realm="asterisk", nonce="1643cd23", uri="sip:192.168.1.11:5060", response="14f189611f61006e1d9539241fb111d1", algorithm=md5 Content-Type: application/sdp Content-Length:?? 269 v=0 o=- 2215885361 2215885361 IN IP4 192.168.1.200 s=pjmedia c=IN IP4 192.168.1.200 t=0 0 m=audio 11000 RTP/AVP 0 8 101 a=rtcp:11001 IN IP4 192.168.1.200 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ?19:42:41.373?? pjsua_core.c RX 433 bytes Response msg 404/INVITE/cseq=846930887 (rdata0x1047941c) from UDP 192.168.1.11:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKPj00ee0000001f6b8b4567;received=192.168.1.200;rport=5060 From: "Freescale1" <sip:5001@192.168.1.11>;tag=00ee0000001c6b8b4567 To: sip:5002 at 192.168.1.11;tag=as4b230045 Call-ID: 00ee0000001d6b8b4567 CSeq: 846930887 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ###################################################################################################### ? ? Please get me some suggestion. ? Thank you so much! ? ? Best Regards Michael Liu ? Email: skysoshy at msn.com dongdong27 at 163.com ________________________________ ????? Windows Live Messenger ???????? ????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080626/85e39b9f/attachment.html
