rendered paste body -- Executing Dial("SIP/511-6b95", "SIP/510&|15|tr") in new stack
We're at 192.168.2.35 port 17502
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x1 (g723)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
Answering with capability 0x10 (g726)
Answering with capability 0x20 (adpcm)
Answering with capability 0x40 (slin)
Answering with capability 0x80 (lpc10)
Answering with capability 0x100 (g729)
Answering with capability 0x200 (speex)
Answering with capability 0x400 (ilbc)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 20 lines
Reliably Transmitting:
INVITE sip:510@192.168.2.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>
Contact: <sip:511@192.168.2.35>
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 08 Nov 2005 22:38:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 471
v=0
o=root 16703 16703 IN IP4 192.168.2.35
s=session
c=IN IP4 192.168.2.35
t=0 0
m=audio 17502 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 192.168.2.8:5060
-- Called 510
Reliably Transmitting:
CANCEL sip:510@192.168.2.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>
Contact: <sip:511@192.168.2.35>
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 192.168.2.8:5060
Scheduling destruction of call '4ccb7fac59aa7096535b37a31297c44f@192.168.2.35' in 15000 ms
-- Executing GotoIf("SIP/511-6b95", "0?s-|1") in new stack
-- Executing GotoIf("SIP/511-6b95", "1?s-|1") in new stack
-- Goto (macro-exten-vm,s-,1)
-- Executing Congestion("SIP/511-6b95", "") in new stack
plasma*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>;tag=339F9BCC-5E1B5319
CSeq: 102 INVITE
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
Contact: <sip:510@192.168.2.8>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1
Content-Length: 0
9 headers, 0 lines
plasma*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>
CSeq: 102 CANCEL
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
Contact: <sip:510@192.168.2.8>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1
Content-Length: 0
9 headers, 0 lines
plasma*CLI>
Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>;tag=339F9BCC-5E1B5319
CSeq: 102 INVITE
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
Contact: <sip:510@192.168.2.8>
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1
Content-Length: 0
9 headers, 0 lines
Transmitting:
ACK sip:510@192.168.2.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK5f2262b4;rport
From: "EngCom2" <sip:511@192.168.2.35>;tag=as748e9b13
To: <sip:510@192.168.2.8>;tag=339F9BCC-5E1B5319
Contact: <sip:511@192.168.2.35>
Call-ID: 4ccb7fac59aa7096535b37a31297c44f@192.168.2.35
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 192.168.2.8:5060
Destroying call '4ccb7fac59aa7096535b37a31297c44f@192.168.2.35'
== Spawn extension (macro-exten-vm, s-, 1) exited non-zero on 'SIP/511-6b95' in macro 'exten-vm'
== Spawn extension (ext-local, 510, 1) exited non-zero on 'SIP/511-6b95'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:510@192.168.2.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK10383d5f
From: "Unknown" <sip:Unknown@192.168.2.35>;tag=as13b74195
To: <sip:510@192.168.2.8>
Contact: <sip:Unknown@192.168.2.35>
Call-ID: 38a66f0705960a4f24625f681250df9d@192.168.2.35
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 08 Nov 2005 22:38:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 192.168.2.8:5060
plasma*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.35:5060;branch=z9hG4bK10383d5f
From: "Unknown" <sip:Unknown@192.168.2.35>;tag=as13b74195
To: <sip:510@192.168.2.8>;tag=18FA6F35-98F9901A
CSeq: 102 OPTIONS
Call-ID: 38a66f0705960a4f24625f681250df9d@192.168.2.35
Contact: <sip:510@192.168.2.8>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.4.1
Content-Length: 0
10 headers, 0 lines
Destroying call '38a66f0705960a4f24625f681250df9d@192.168.2.35'