On 06/21/09 19:23, Tzafrir Cohen wrote: >On Sun, Jun 21, 2009 at 09:28:21AM -0600, Joseph wrote: > >> No, it does not wait 20sec. after the first ring it goes directly to >> voicemail (so a second or two). >> The strange part is that when I call the same extensions from PSTN >> line it rings 20 sec. so about 3 or 4 rings; it only happens when I >> try to call internally from one extension to another. > >If you need more data, how about sip debug? >
I don't see anything unusual, here is a call with sip debug enabled from ext. 11 to ext. 711 <-------------> syscon4*CLI> <--- SIP read from 10.0.0.102:5064 ---> INVITE sip:7...@10.0.0.109 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-bf82100b From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109> Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:1...@10.0.0.102:5064> Expires: 240 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 4821148 4821148 IN IP4 10.0.0.102 s=- c=IN IP4 10.0.0.102 t=0 0 m=audio 16404 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/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 <-------------> --- (14 headers 19 lines) --- Sending to 10.0.0.102 : 5064 (no NAT) Using INVITE request as basis request - e8487bdc-a19f1...@10.0.0.102 <--- Reliably Transmitting (no NAT) to 10.0.0.102:5064 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-bf82100b;received=10.0.0.102 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as3085a901 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 101 INVITE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a45dcdb" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e8487bdc-a19f1...@10.0.0.102' in 32000 ms (Method: INVITE) Found user '11' <--- SIP read from 10.0.0.102:5064 ---> ACK sip:7...@10.0.0.109 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-bf82100b From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as3085a901 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 101 ACK Max-Forwards: 70 Contact: <sip:1...@10.0.0.102:5064> User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from 10.0.0.102:5064 ---> INVITE sip:7...@10.0.0.109 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-2e6fd9e7 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109> Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="11",realm="asterisk",nonce="1a45dcdb",uri="sip:7...@10.0.0.109",algorithm=MD5,response="d3751874a3aa3de10fa5cebf958388ce" Contact: <sip:1...@10.0.0.102:5064> Expires: 240 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 4821148 4821148 IN IP4 10.0.0.102 s=- c=IN IP4 10.0.0.102 t=0 0 m=audio 16404 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/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 <-------------> --- (15 headers 19 lines) --- Sending to 10.0.0.102 : 5064 (no NAT) Using INVITE request as basis request - e8487bdc-a19f1...@10.0.0.102 Found user '11' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.102:16404 Found audio description format PCMU for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format PCMA for ID 8 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.0.102:16404 Looking for 711 in internal (domain 10.0.0.109) list_route: hop: <sip:1...@10.0.0.102:5064> <--- Transmitting (no NAT) to 10.0.0.102:5064 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-2e6fd9e7;received=10.0.0.102 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109> Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 102 INVITE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:7...@10.0.0.109> Content-Length: 0 <------------> -- Executing [...@internal:1] Dial("SIP/11-007a8000", "SIP/711|30|rw") in new stack Audio is at 10.0.0.109 port 19092 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.111:5068: INVITE sip:7...@10.0.0.111:5068 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5;rport From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 To: <sip:7...@10.0.0.111:5068> Contact: <sip:1...@10.0.0.109> Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 INVITE User-Agent: Centrala Max-Forwards: 70 Date: Sun, 21 Jun 2009 16:48:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 260 v=0 o=root 32459 32459 IN IP4 10.0.0.109 s=session c=IN IP4 10.0.0.109 t=0 0 m=audio 19092 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/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 --- -- Called 711 <--- Transmitting (no NAT) to 10.0.0.102:5064 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-2e6fd9e7;received=10.0.0.102 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as0e5759d0 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 102 INVITE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:7...@10.0.0.109> Content-Length: 0 <------------> syscon4*CLI> <--- SIP read from 10.0.0.111:5068 ---> SIP/2.0 100 Trying To: <sip:7...@10.0.0.111:5068> From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5 Server: Sipura/SPA2002-3.1.2(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 10.0.0.111:5068 ---> SIP/2.0 180 Ringing To: <sip:7...@10.0.0.111:5068>;tag=44fef0f952feffd3i1 From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5 Server: Sipura/SPA2002-3.1.2(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/711-007ac340 is ringing -- Nobody picked up in 30000 ms Scheduling destruction of SIP dialog '01ac7e4f6958ab3609efb9d70d467...@10.0.0.109' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 10.0.0.111:5068: CANCEL sip:7...@10.0.0.111:5068 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5;rport From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 To: <sip:7...@10.0.0.111:5068> Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 CANCEL User-Agent: Centrala Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '01ac7e4f6958ab3609efb9d70d467...@10.0.0.109' in 32000 ms (Method: INVITE) -- Executing [...@internal:2] VoiceMail("SIP/11-007a8000", "u711") in new stack Audio is at 10.0.0.109 port 18860 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.0.0.102:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-2e6fd9e7;received=10.0.0.102 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as0e5759d0 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 102 INVITE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:7...@10.0.0.109> Content-Type: application/sdp Content-Length: 260 v=0 o=root 32459 32459 IN IP4 10.0.0.109 s=session c=IN IP4 10.0.0.109 t=0 0 m=audio 18860 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/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 <------------> -- <SIP/11-007a8000> Playing 'vm-theperson' (language 'en') syscon4*CLI> <--- SIP read from 10.0.0.111:5068 ---> SIP/2.0 487 Request Terminated To: <sip:7...@10.0.0.111:5068>;tag=44fef0f952feffd3i1 From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5 Server: Sipura/SPA2002-3.1.2(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 10.0.0.111:5068: ACK sip:7...@10.0.0.111:5068 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5;rport From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 To: <sip:7...@10.0.0.111:5068>;tag=44fef0f952feffd3i1 Contact: <sip:1...@10.0.0.109> Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 ACK User-Agent: Centrala Max-Forwards: 70 Content-Length: 0 --- syscon4*CLI> <--- SIP read from 10.0.0.111:5068 ---> SIP/2.0 200 OK To: <sip:7...@10.0.0.111:5068>;tag=44fef0f952feffd3i1 From: "Joseph" <sip:1...@10.0.0.109>;tag=as30825c41 Call-ID: 01ac7e4f6958ab3609efb9d70d467...@10.0.0.109 CSeq: 102 CANCEL Via: SIP/2.0/UDP 10.0.0.109:5060;branch=z9hG4bK723eaac5 Server: Sipura/SPA2002-3.1.2(a) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '01ac7e4f6958ab3609efb9d70d467...@10.0.0.109' Method: INVITE <--- SIP read from 10.0.0.102:5064 ---> ACK sip:7...@10.0.0.109 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-e3cedcb4 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as0e5759d0 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="11",realm="asterisk",nonce="1a45dcdb",uri="sip:7...@10.0.0.109",algorithm=MD5,response="d2fd92784cf29e51c0a0e270a462e3b9" Contact: <sip:1...@10.0.0.102:5064> User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- <SIP/11-007a8000> Playing 'digits/7' (language 'en') -- <SIP/11-007a8000> Playing 'digits/1' (language 'en') -- <SIP/11-007a8000> Playing 'digits/1' (language 'en') syscon4*CLI> <--- SIP read from 10.0.0.102:5064 ---> BYE sip:7...@10.0.0.109 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-dd947196 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as0e5759d0 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="11",realm="asterisk",nonce="1a45dcdb",uri="sip:7...@10.0.0.109",algorithm=MD5,response="747fa5968012da031745d945cc0bbc69" User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 10.0.0.102 : 5064 (no NAT) <--- Transmitting (no NAT) to 10.0.0.102:5064 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.102:5064;branch=z9hG4bK-dd947196;received=10.0.0.102 From: <sip:1...@10.0.0.109>;tag=4a2d66f8f44bc883o0 To: <sip:7...@10.0.0.109>;tag=as0e5759d0 Call-ID: e8487bdc-a19f1...@10.0.0.102 CSeq: 103 BYE User-Agent: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:7...@10.0.0.109> Content-Length: 0 -- Joseph _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users