I have a strange problem. I'm using the same dialplan to call 1800-number:
[toll-free] ;second "7" audiocodes strips exten => _71800XXXXXXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr) When I call this number (through pstn-5665) 18005000347 the phone always rings busy. When I call any other 1800-number the calls goes through. When I call the same phone number 18005000347 through a different line the calls goes through every time. Here is call (busy) trace to that 18005000347 with sip debug ON: Can anybody decipher why I'm getting busy signal to that particular 1800-number but not others? <--- SIP read from UDP:10.0.0.110:5060 ---> OPTIONS sip:gateway@10.0.0.110 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404 Max-Forwards: 70 From: <sip:gateway@10.0.0.110:5060>;tag=1c1457828994 To: <sip:gateway@10.0.0.110> Call-ID: 1457828497512012183855@10.0.0.110 CSeq: 1 OPTIONS Contact: <sip:gateway@10.0.0.110:5060> Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Accept: application/sdp, application/simple-message-summary, message/sipfrag Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Looking for gateway in default (domain 10.0.0.110) <--- Transmitting (NAT) to 10.0.0.110:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060 From: <sip:gateway@10.0.0.110:5060>;tag=1c1457828994 To: <sip:gateway@10.0.0.110>;tag=as7091ae01 Call-ID: 1457828497512012183855@10.0.0.110 CSeq: 1 OPTIONS Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in 32000 ms (Method: OPTIONS) Reliably Transmitting (no NAT) to 81.15.150.20:5060: OPTIONS sip:sip.actio.pl SIP/2.0 Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5 Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.0.0.100>;tag=as64f6417c To: <sip:sip.actio.pl> Contact: <sip:asterisk@10.0.0.100:5060> Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060 CSeq: 102 OPTIONS User-Agent: Centrala Date: Fri, 06 Jan 2012 01:39:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:81.15.150.20:5060 ---> SIP/2.0 501 Unsupported Method Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715 To: <sip:sip.actio.pl>;tag=4fc8ac12 From: "asterisk"<sip:asterisk@10.0.0.100>;tag=as64f6417c Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '66070317301f64861df62d20769ba385@10.0.0.100:5060' Method: OPTIONS -- Accepted AUTHENTICATED TBD call from 10.0.0.108 <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360 Max-Forwards: 70 From: <sip:11@10.0.0.110>;tag=1c1472330741 To: <sip:11@10.0.0.110> Call-ID: 809487713120129287@10.0.0.110 CSeq: 245 REGISTER Contact: <sip:11@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.0.0.110:5060 (NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110 From: <sip:11@10.0.0.110>;tag=1c1472330741 To: <sip:11@10.0.0.110>;tag=as21c548bd Call-ID: 809487713120129287@10.0.0.110 CSeq: 245 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a451a5b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428 Max-Forwards: 70 From: <sip:11@10.0.0.110>;tag=1c1472330741 To: <sip:11@10.0.0.110> Call-ID: 809487713120129287@10.0.0.110 CSeq: 246 REGISTER Authorization: Digest username="11",realm="asterisk",nonce="3a451a5b",uri="sip:10.0.0.100",algorithm=MD5,response="5dd6df18064f3d23cb86ca306820e596" Contact: <sip:11@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.0.0.110:5060 (no NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428;received=10.0.0.110 From: <sip:11@10.0.0.110>;tag=1c1472330741 To: <sip:11@10.0.0.110>;tag=as21c548bd Call-ID: 809487713120129287@10.0.0.110 CSeq: 246 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:11@10.0.0.110:5060>;expires=3600 Date: Fri, 06 Jan 2012 01:39:11 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1473585078 Max-Forwards: 70 From: <sip:369@10.0.0.110>;tag=1c1473580481 To: <sip:369@10.0.0.110> Call-ID: 809480513120129287@10.0.0.110 CSeq: 245 REGISTER Contact: <sip:369@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.0.0.110:5060 (NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1473585078;received=10.0.0.110 From: <sip:369@10.0.0.110>;tag=1c1473580481 To: <sip:369@10.0.0.110>;tag=as5e849b38 Call-ID: 809480513120129287@10.0.0.110 CSeq: 245 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4990d6db" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809480513120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1473657330 Max-Forwards: 70 From: <sip:369@10.0.0.110>;tag=1c1473580481 To: <sip:369@10.0.0.110> Call-ID: 809480513120129287@10.0.0.110 CSeq: 246 REGISTER Authorization: Digest username="369",realm="asterisk",nonce="4990d6db",uri="sip:10.0.0.100",algorithm=MD5,response="2040330b2174b1bf6fa6b5270f2045da" Contact: <sip:369@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.0.0.110:5060 (no NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1473657330;received=10.0.0.110 From: <sip:369@10.0.0.110>;tag=1c1473580481 To: <sip:369@10.0.0.110>;tag=as5e849b38 Call-ID: 809480513120129287@10.0.0.110 CSeq: 246 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:369@10.0.0.110:5060>;expires=3600 Date: Fri, 06 Jan 2012 01:39:12 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809480513120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:81.15.150.20:5060 ---> <-------------> -- Accepting DIAL from 10.0.0.108, formats = 0x4 (ulaw) -- Executing [718005000347@internal:1] Dial("IAX2/iaxy-322-2562", "SIP/718005000347@pstn-5665,60,tr") in new stack == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Audio is at 5060 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 10.0.0.110:5060: INVITE sip:718005000347@10.0.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK37dc4d56 Max-Forwards: 70 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060> Contact: <sip:322@10.0.0.100:5060> Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 102 INVITE User-Agent: Centrala Date: Fri, 06 Jan 2012 01:39:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 283 v=0 o=root 1953303792 1953303792 IN IP4 10.0.0.100 s=Asterisk PBX 1.8.7.2 c=IN IP4 10.0.0.100 t=0 0 m=audio 19756 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 SIP/718005000347@pstn-5665 <--- SIP read from UDP:10.0.0.110:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK37dc4d56 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 102 INVITE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.0.0.110:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK37dc4d56 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 102 INVITE Contact: <sip:pstn-5665@10.0.0.110:5060> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Type: application/sdp Content-Length: 250 v=0 o=AudiocodesGW 1483062808 1483062678 IN IP4 10.0.0.110 s=Phone-Call c=IN IP4 10.0.0.110 t=0 0 m=audio 6000 RTP/AVP 0 101 c=IN IP4 10.0.0.110 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.0.110:6000 -- SIP/pstn-5665-0000002c is making progress passing it to IAX2/iaxy-322-2562 <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1485714873 Max-Forwards: 70 From: <sip:pstn-5665@10.0.0.110>;tag=1c1485710213 To: <sip:pstn-5665@10.0.0.110> Call-ID: 809465873120129287@10.0.0.110 CSeq: 245 REGISTER Contact: <sip:pstn-5665@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.0.0.110:5060 (NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1485714873;received=10.0.0.110 From: <sip:pstn-5665@10.0.0.110>;tag=1c1485710213 To: <sip:pstn-5665@10.0.0.110>;tag=as5cc9098f Call-ID: 809465873120129287@10.0.0.110 CSeq: 245 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d3d7182" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809465873120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1485785133 Max-Forwards: 70 From: <sip:pstn-5665@10.0.0.110>;tag=1c1485710213 To: <sip:pstn-5665@10.0.0.110> Call-ID: 809465873120129287@10.0.0.110 CSeq: 246 REGISTER Authorization: Digest username="pstn-5665",realm="asterisk",nonce="1d3d7182",uri="sip:10.0.0.100",algorithm=MD5,response="ae65d593d082a7da70ddb6a3cc049070" Contact: <sip:pstn-5665@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.0.0.110:5060 (no NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1485785133;received=10.0.0.110 From: <sip:pstn-5665@10.0.0.110>;tag=1c1485710213 To: <sip:pstn-5665@10.0.0.110>;tag=as5cc9098f Call-ID: 809465873120129287@10.0.0.110 CSeq: 246 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:pstn-5665@10.0.0.110:5060>;expires=3600 Date: Fri, 06 Jan 2012 01:39:17 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809465873120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1487588038 Max-Forwards: 70 From: <sip:pstn-1270@10.0.0.110>;tag=1c1487583344 To: <sip:pstn-1270@10.0.0.110> Call-ID: 809473423120129287@10.0.0.110 CSeq: 245 REGISTER Contact: <sip:pstn-1270@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 10.0.0.110:5060 (NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1487588038;received=10.0.0.110 From: <sip:pstn-1270@10.0.0.110>;tag=1c1487583344 To: <sip:pstn-1270@10.0.0.110>;tag=as65770b20 Call-ID: 809473423120129287@10.0.0.110 CSeq: 245 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20bd8ce7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809473423120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> REGISTER sip:10.0.0.100 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1487657797 Max-Forwards: 70 From: <sip:pstn-1270@10.0.0.110>;tag=1c1487583344 To: <sip:pstn-1270@10.0.0.110> Call-ID: 809473423120129287@10.0.0.110 CSeq: 246 REGISTER Authorization: Digest username="pstn-1270",realm="asterisk",nonce="20bd8ce7",uri="sip:10.0.0.100",algorithm=MD5,response="242b508a659c2b205483cb5af5ff1d1b" Contact: <sip:pstn-1270@10.0.0.110:5060>;expires=3600 Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 3600 User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 10.0.0.110:5060 (no NAT) <--- Transmitting (no NAT) to 10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1487657797;received=10.0.0.110 From: <sip:pstn-1270@10.0.0.110>;tag=1c1487583344 To: <sip:pstn-1270@10.0.0.110>;tag=as65770b20 Call-ID: 809473423120129287@10.0.0.110 CSeq: 246 REGISTER Server: Centrala Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:pstn-1270@10.0.0.110:5060>;expires=3600 Date: Fri, 06 Jan 2012 01:39:17 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '809473423120129287@10.0.0.110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK37dc4d56 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 102 INVITE Contact: <sip:pstn-5665@10.0.0.110:5060> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Type: application/sdp Content-Length: 250 v=0 o=AudiocodesGW 1483062808 1483062678 IN IP4 10.0.0.110 s=Phone-Call c=IN IP4 10.0.0.110 t=0 0 m=audio 6000 RTP/AVP 0 101 c=IN IP4 10.0.0.110 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- list_route: hop: <sip:pstn-5665@10.0.0.110:5060> set_destination: Parsing <sip:pstn-5665@10.0.0.110:5060> for address/port to send to set_destination: set destination to 10.0.0.110:5060 Transmitting (no NAT) to 10.0.0.110:5060: ACK sip:pstn-5665@10.0.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK11054a44 Max-Forwards: 70 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Contact: <sip:322@10.0.0.100:5060> Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 102 ACK User-Agent: Centrala Content-Length: 0 --- -- SIP/pstn-5665-0000002c answered IAX2/iaxy-322-2562 Really destroying SIP dialog 'REGISTER_00D0E9400836_T1185900361@10.0.0.107' Method: REGISTER Scheduling destruction of SIP dialog '3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060' in 32000 ms (Method: INVITE) set_destination: Parsing <sip:pstn-5665@10.0.0.110:5060> for address/port to send to set_destination: set destination to 10.0.0.110:5060 Reliably Transmitting (no NAT) to 10.0.0.110:5060: BYE sip:pstn-5665@10.0.0.110:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK4df9b567 Max-Forwards: 70 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 103 BYE User-Agent: Centrala X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (internal, 718005000347, 1) exited non-zero on 'IAX2/iaxy-322-2562' -- Hungup 'IAX2/iaxy-322-2562' <--- SIP read from UDP:10.0.0.110:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK4df9b567 From: "Joseph :-)" <sip:322@10.0.0.100>;tag=as7ed1ca7a To: <sip:718005000347@10.0.0.110:5060>;tag=1c1483033307 Call-ID: 3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060 CSeq: 103 BYE Contact: <sip:pstn-5665@10.0.0.110:5060> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-/v.5.80A.032.003 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3ddf9e526ede57025eb4916d729ea24d@10.0.0.100:5060' Method: INVITE -- Joseph -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users