What does your extensions.conf look like? Did you answer() the call first ?
wes -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Lee Sent: Monday, February 09, 2004 6:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with Sip call problems - Whats not working? When I press a key (8) on the phone, it should play a few bits of audio and go to voicemail for testing. I dont get any sound back, and it appears the call is progressing without me. Here is the console output with sip debug: Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 287 v=0 o=p3000 5972727 56415 IN IP4 10.10.10.2 s=SIP Call c=IN IP4 10.10.10.2 t=0 0 m=audio 10096 RTP/AVP 0 8 18 4 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 10.10.10.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format pcmu Found description format pcma Found description format G729 Found description format g723 Found description format g726 Found description format telephone-event Capabilities: us - 14, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 8 in wellingborough-road list_route: hop: <sip:[EMAIL PROTECTED]> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 10.10.10.2:5060 -- Executing BackGround("SIP/p3000-1186", "sounds/carried-away-by-monkeys") in new stack We're at 10.10.10.3 port 17190 Answering with preferred capability 2 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 -- Playing 'sounds/carried-away-by-monkeys' (language 'en') babybell*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 287 v=0 o=p3000 5972727 56415 IN IP4 10.10.10.2 s=SIP Call c=IN IP4 10.10.10.2 t=0 0 m=audio 10096 RTP/AVP 0 8 18 4 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Ignoring this request We're at 10.10.10.3 port 17190 Answering with preferred capability 2 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17879 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 babybell*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-AAE-26994 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 9 headers, 0 lines Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 -- Executing BackGround("SIP/p3000-1186", "sounds/lots-o-monkeys") in new stack -- Playing 'sounds/lots-o-monkeys' (language 'en') Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 To: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 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 to 10.10.10.2:5060 Feb 9 09:47:15 WARNING[81926]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) == Spawn extension (wellingborough-road, 8, 2) exited non-zero on 'SIP/p3000-1186' -- Executing Hangup("SIP/p3000-1186", "") in new stack == Spawn extension (wellingborough-road, h, 1) exited non-zero on 'SIP/p3000-1186' set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to 10.10.10.2, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d From: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 To: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.10.10.2:5060 babybell*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d From: <sip:[EMAIL PROTECTED]>;tag=as3bf9fee8 To: p3000 <sip:[EMAIL PROTECTED]:5060>;tag=TdR-16808 Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE Content-Length: 0 7 headers, 0 lines Message is BYE ####################### Calls originating at FXO and going to this extension work fine. Calls originating at this extension are a problem. Any help would be great Regards Chris Lee _______________________________________________ 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 _______________________________________________ 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
