[asterisk-users] DTMF

[David @ULC]

Any idea whats wrong ?

On Fri, Feb 20, 2009 at 2:32 AM, David @ULC <ucoms2001 at gmail.com> wrote:

>
> --- (12 headers 0 lines) ---
> Sending to 192.168.0.50 : 12714 (NAT)
> Transmitting (NAT) to 192.168.0.50:12714:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.50:12714
> ;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714
> From: "cc106"<sip:cc106 at 192.168.0.2 <sip%3Acc106 at 192.168.0.2>
> >;tag=7f1cff22
> To: "817275691533"<sip:817275691533 at 192.168.0.2<sip%3A817275691533 at 192.168.0.2>
> >;tag=as02559696
> Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.
> CSeq: 3 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:817275691533 at 192.168.0.2 <sip%3A817275691533 at 192.168.0.2>>
> Content-Length: 0
> X-Asterisk-HangupCause: Normal Clearing
>
>
>
> ---
> Scheduling destruction of call
> '617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx' in 32000 ms
> set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
> address/port to send to
> set_destination: set destination to 8.14.xxx.xxx, port 5060
> Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060:
> BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport
> From: "cc106" <sip:fiddialer at 59.xxx.xx.xx>;tag=as3f9466a7
> To: <sip:17275691533 at 8.14.xxx.xxx>;tag=1902000923108720995156225
> Call-ID: 617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>   == Spawn extension (default, 817275691533, 2) exited non-zero on
> 'SIP/cc106-b7a1a9d0'
>     -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://
> 127.0.0.1:4577/call_log") in new stack
>     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
>     -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://
> 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)")
> in new stack
>     -- AGI Script agi://
> 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)
> completed, returning 0
> Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.'
> vicidialnow*CLI>
> <-- SIP read from 8.14.xxx.xxx:5060:
> SIP/2.0 200 OK
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport
> From: "V0219160007000134649" <sip:fiddialer at 59.xxx.xx.xx>;tag=as79fae976
> Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
> To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
> Contact: <sip:8.14.xxx.xxx:5060;transport=udp>
> Content-Type: application/sdp
> Content-Length: 225
>
> v=0
> o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx
> s=VoipSIP
> i=Audio Session
> c=IN IP4 8.14.xxx.xxx
> t=0 0
> m=audio 6220 RTP/AVP 18 101
> a=rtpmap:18 G729/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> --- (9 headers 11 lines) ---
> Found RTP audio format 18
> Found RTP audio format 101
> Peer audio RTP is at port 8.14.xxx.xxx:6220
> Found description format G729
> Found description format telephone-event
> Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0
> (nothing), combined - 0x100 (g729)
> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp>
> set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
> address/port to send to
> set_destination: set destination to 8.14.xxx.xxx, port 5060
> Transmitting (no NAT) to 8.14.xxx.xxx:5060:
> ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport
> From: "V0219160007000134649" <sip:fiddialer at 59.xxx.xx.xx>;tag=as79fae976
> To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
> Contact: <sip:fiddialer at 59.xxx.xx.xx>
> Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
>
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