Hi Stefan, Thank you very much for your quick response :) I tried again with r1434 but I'm getting some extrange error. By looking at SIP captures, I see the INVITE is generated withoud SDP...
# U +5.653972 192.168.1.112:5060 -> 192.168.1.115:5080 INVITE sip:[email protected]:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as51b25bdf To: <sip:[email protected]:5080> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9 Date: Thu, 11 Jun 2009 22:24:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 208 v=0 o=root 1124002214 1124002214 IN IP4 192.168.1.112 s=Asterisk PBX 1.6.0.9 c=IN IP4 192.168.1.112 t=0 0 m=audio 8464 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv # U +0.018650 192.168.1.115:5080 -> 192.168.1.112:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport From: "asterisk" <sip:[email protected]>;tag=as51b25bdf To: <sip:[email protected]:5080>;tag=361D04EE-4A31842D000ED85A-B770FB90 Call-ID: [email protected] CSeq: 102 INVITE Server: Sip Express Media Server (1.1.0-dev-r1434M (i386/linux)) Contact: <sip:[email protected]:5080> Content-Length: 0 # U +0.000656 192.168.1.115:5080 -> 192.168.1.112:5060 SIP/2.0 488 could not find compatible payload Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK55596cc3;rport From: "asterisk" <sip:[email protected]>;tag=as51b25bdf To: <sip:[email protected]:5080>;tag=67D6E165-4A31842D000EDBFF-B75E8B90 Call-ID: [email protected] CSeq: 102 INVITE Server: Sip Express Media Server (1.1.0-dev-r1434M (i386/linux)) Content-Length: 0 # U +0.082672 192.168.1.115:5080 -> 213.192.59.75:5060 INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5080;branch=z9hG4bKUWp6haKu From: "asterisk" <sip:[email protected]>;tag=1FF17244-4A31842D000EDA14-B75E8B90 To: sip:[email protected] CSeq: 10 INVITE Call-ID: [email protected] Contact: <sip:[email protected]:5080> Content-Length: 0 # U +0.081238 213.192.59.75:5060 -> 192.168.1.115:5080 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.115:5080;branch=z9hG4bKUWp6haKu;rport=5080;received=85.84.127.147 From: "asterisk" <sip:[email protected]>;tag=1FF17244-4A31842D000EDA14-B75E8B90 To: sip:[email protected] CSeq: 10 INVITE Call-ID: [email protected] Server: Sip EXpress router (2.1.0-dev23-make (i386/linux)) Content-Length: 0 Warning: 392 213.192.59.75:5060 "Noisy feedback tells: pid=2613 req_src_ip=85.84.127.147 req_src_port=5080 in_uri=sip:[email protected] out_uri=sip:[email protected]:5074 via_cnt==1" I'm also a little confused about how the dsm is executed... can you explain me with more detail when on invite and when on session start are lanuched please? Thanks in advance! Regards, PD: Find attached the dsm file. I made changes according to your suggestions, buy I may be missing something... Thanks again! :) -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/
x_b2bua.dsm
Description: Binary data
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