It took 36 seconds for that number to answer when I called it and it looks like the call hung up after 32000 ms when you dialed via asterisk. -- Jim Dickenson mailto:dicken...@cfmc.com
CfMC http://www.cfmc.com/ On Jan 5, 2012, at 5:45 PM, Joseph wrote: > 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 -- _____________________________________________________________________ -- 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
