[asterisk-users] calling specific 1800-number not going through.
Jim Dickenson
dickenson at cfmc.com
Thu Jan 5 19:55:11 CST 2012
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:dickenson at 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 at 10.0.0.110 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
> Max-Forwards: 70
> From: <sip:gateway at 10.0.0.110:5060>;tag=1c1457828994
> To: <sip:gateway at 10.0.0.110>
> Call-ID: 1457828497512012183855 at 10.0.0.110
> CSeq: 1 OPTIONS
> Contact: <sip:gateway at 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 at 10.0.0.110:5060>;tag=1c1457828994
> To: <sip:gateway at 10.0.0.110>;tag=as7091ae01
> Call-ID: 1457828497512012183855 at 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 at 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 at 10.0.0.100>;tag=as64f6417c
> To: <sip:sip.actio.pl>
> Contact: <sip:asterisk at 10.0.0.100:5060>
> Call-ID: 66070317301f64861df62d20769ba385 at 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 at 10.0.0.100>;tag=as64f6417c
> Call-ID: 66070317301f64861df62d20769ba385 at 10.0.0.100:5060
> CSeq: 102 OPTIONS
> Content-Length: 0
>
> <------------->
> --- (7 headers 0 lines) ---
> Really destroying SIP dialog '66070317301f64861df62d20769ba385 at 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 at 10.0.0.110>;tag=1c1472330741
> To: <sip:11 at 10.0.0.110>
> Call-ID: 809487713120129287 at 10.0.0.110
> CSeq: 245 REGISTER
> Contact: <sip:11 at 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 at 10.0.0.110>;tag=1c1472330741
> To: <sip:11 at 10.0.0.110>;tag=as21c548bd
> Call-ID: 809487713120129287 at 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 at 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 at 10.0.0.110>;tag=1c1472330741
> To: <sip:11 at 10.0.0.110>
> Call-ID: 809487713120129287 at 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 at 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 at 10.0.0.110>;tag=1c1472330741
> To: <sip:11 at 10.0.0.110>;tag=as21c548bd
> Call-ID: 809487713120129287 at 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 at 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 at 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 at 10.0.0.110>;tag=1c1473580481
> To: <sip:369 at 10.0.0.110>
> Call-ID: 809480513120129287 at 10.0.0.110
> CSeq: 245 REGISTER
> Contact: <sip:369 at 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 at 10.0.0.110>;tag=1c1473580481
> To: <sip:369 at 10.0.0.110>;tag=as5e849b38
> Call-ID: 809480513120129287 at 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 at 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 at 10.0.0.110>;tag=1c1473580481
> To: <sip:369 at 10.0.0.110>
> Call-ID: 809480513120129287 at 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 at 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 at 10.0.0.110>;tag=1c1473580481
> To: <sip:369 at 10.0.0.110>;tag=as5e849b38
> Call-ID: 809480513120129287 at 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 at 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 at 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 at internal:1] Dial("IAX2/iaxy-322-2562", "SIP/718005000347 at 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 at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>
> Contact: <sip:322 at 10.0.0.100:5060>
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 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 at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 10.0.0.100:5060
> CSeq: 102 INVITE
> Contact: <sip:pstn-5665 at 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 at 10.0.0.110>;tag=1c1485710213
> To: <sip:pstn-5665 at 10.0.0.110>
> Call-ID: 809465873120129287 at 10.0.0.110
> CSeq: 245 REGISTER
> Contact: <sip:pstn-5665 at 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 at 10.0.0.110>;tag=1c1485710213
> To: <sip:pstn-5665 at 10.0.0.110>;tag=as5cc9098f
> Call-ID: 809465873120129287 at 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 at 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 at 10.0.0.110>;tag=1c1485710213
> To: <sip:pstn-5665 at 10.0.0.110>
> Call-ID: 809465873120129287 at 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 at 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 at 10.0.0.110>;tag=1c1485710213
> To: <sip:pstn-5665 at 10.0.0.110>;tag=as5cc9098f
> Call-ID: 809465873120129287 at 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 at 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 at 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 at 10.0.0.110>;tag=1c1487583344
> To: <sip:pstn-1270 at 10.0.0.110>
> Call-ID: 809473423120129287 at 10.0.0.110
> CSeq: 245 REGISTER
> Contact: <sip:pstn-1270 at 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 at 10.0.0.110>;tag=1c1487583344
> To: <sip:pstn-1270 at 10.0.0.110>;tag=as65770b20
> Call-ID: 809473423120129287 at 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 at 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 at 10.0.0.110>;tag=1c1487583344
> To: <sip:pstn-1270 at 10.0.0.110>
> Call-ID: 809473423120129287 at 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 at 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 at 10.0.0.110>;tag=1c1487583344
> To: <sip:pstn-1270 at 10.0.0.110>;tag=as65770b20
> Call-ID: 809473423120129287 at 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 at 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 at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 10.0.0.100:5060
> CSeq: 102 INVITE
> Contact: <sip:pstn-5665 at 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 at 10.0.0.110:5060>
> set_destination: Parsing <sip:pstn-5665 at 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 at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Contact: <sip:322 at 10.0.0.100:5060>
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 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 at 10.0.0.107' Method: REGISTER
> Scheduling destruction of SIP dialog '3ddf9e526ede57025eb4916d729ea24d at 10.0.0.100:5060' in 32000 ms (Method: INVITE)
> set_destination: Parsing <sip:pstn-5665 at 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 at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 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 at 10.0.0.100>;tag=as7ed1ca7a
> To: <sip:718005000347 at 10.0.0.110:5060>;tag=1c1483033307
> Call-ID: 3ddf9e526ede57025eb4916d729ea24d at 10.0.0.100:5060
> CSeq: 103 BYE
> Contact: <sip:pstn-5665 at 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 at 10.0.0.100:5060' Method: INVITE
>
> --
> Joseph
>
> --
> _____________________________________________________________________
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