Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my endpoint. I think this is not normal (or maybe yes?). I'am attaching my extension.conf and my sip.conf.
extension.conf [general] static=yes writeprotect=no [OUTGOING] ;exten => _00562.,1,Dial(SIP/[EMAIL PROTECTED]) exten => _00569.,1,Dial(SIP/[EMAIL PROTECTED]) ----------------------------------- sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;context = sip ;default Default for incoming calls context=OUTGOING autocreatepeer=yes --------------------------- Here is part of the debug from Asterisk (sip debug ip (xxx.xxx.148.246)) We're at xxx.xxx.148.232 port 19848 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.148.246;branch=0 Via: SIP/2.0/UDP xxx.xxx.148.242:5060;branch=z9hG4bK2142c11da4177 Record-Route: <sip:[EMAIL PROTECTED];ftag=2142c11da4;lr=on> From: <sip:[EMAIL PROTECTED]>;tag=2142c11da4 To: <sip:[EMAIL PROTECTED]>;tag=as1be17fe7 Call-ID: [EMAIL PROTECTED] CSeq: 177 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 265 v=0 o=root 13611 13611 IN IP4 xxx.xxx.148.232 s=session c=IN IP4 xxx.xxx.148.232 t=0 0 m=audio 19848 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to xxx.xxx.148.246:5060 sipquest*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: <sip:[EMAIL PROTECTED];ftag=2142c11da4;lr=on> Via: SIP/2.0/UDP xxx.xxx.148.246;branch=0 Via: SIP/2.0/UDP xxx.xxx.148.242:5060;branch=z9hG4bK2142c11da4177 From: <sip:[EMAIL PROTECTED]>;tag=2142c11da4 To: <sip:[EMAIL PROTECTED]>;tag=as1be17fe7 Call-ID: [EMAIL PROTECTED] CSeq: 177 ACK Content-Length: 0 Max-Forwards: 69 10 headers, 0 lines set_destination: Parsing <sip:[EMAIL PROTECTED];ftag=2142c11da4;lr=on> for address/port to send to set_destination: set destination to xxx.xxx.148.246, port 5060 We're at xxx.xxx.148.232 port 19848 Answering with capability 0x4(ULAW) Answering with non-codec capability 0x1(G723) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport Route: <sip:[EMAIL PROTECTED]> From: <sip:[EMAIL PROTECTED]>;tag=as1be17fe7 To: <sip:[EMAIL PROTECTED]>;tag=2142c11da4 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 216 v=0 o=root 13611 13612 IN IP4 xxx.xxx.154.50 s=session c=IN IP4 xxx.xxx.154.50 t=0 0 m=audio 24670 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 =silenceSupp:off - - - - (no NAT) to xxx.xxx.148.246:5060 sipquest*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport=5060 From: <sip:[EMAIL PROTECTED]>;tag=as1be17fe7 To: <sip:[EMAIL PROTECTED]>;tag=2142c11da4 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 xxx.xxx.148.246:5060 "Noisy feedback tells: pid=26918 req_src_ip=xxx.xxx.148.232 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri=sip:[EMAIL PROTECTED] via_cnt==1" 9 headers, 0 lines sipquest*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport=5060 From: <sip:[EMAIL PROTECTED]>;tag=as1be17fe7 To: <sip:[EMAIL PROTECTED]>;tag=2142c11da4 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 207 v=0 o=5555832351 1114766125 1114766125 IN IP4 xxx.xxx.148.242 s=AddPac Gateway SDP c=IN IP4 xxx.xxx.148.242 t=1114766125 0 m=audio 23268 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 10 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port xxx.xxx.148.242:23268 Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: <sip:[EMAIL PROTECTED]> set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to xxx.xxx.148.242, port 5060 Hope that someone can help me. Thanks in advance. Best Regards Ricardo Martinez _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users