Here are the log files for the calls. These are internal calls, from one extension to the other.
Calling from the broken extension (this part works):
asteriskCLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
Really destroying SIP dialog ‘64184ffd380cf2dc21ac76aa67eebd97@10.0.0.1’ Method: OPTIONS
asteriskCLI> sip set debug peer 3504
SIP Debugging Enabled for IP: 10.0.0.178:5060
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
OPTIONS sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6f3da475;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as20a71b6d
To: sip:3504@10.0.0.178
Contact: sip:asterisk@10.0.0.1
Call-ID: 0ad7412275d08412275aaa51006b4f71@10.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:57:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6f3da475;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as20a71b6d
To: sip:3504@10.0.0.178;tag=31C44624-21E00673
CSeq: 102 OPTIONS
Call-ID: 0ad7412275d08412275aaa51006b4f71@10.0.0.1
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘0ad7412275d08412275aaa51006b4f71@10.0.0.1’ Method: OPTIONS
– Executing [3504@main:1] Answer(“SIP/4999-0831e6d0”, “”) in new stack
– Executing [3504@main:2] GotoIf(“SIP/4999-0831e6d0”, “0?reset_name:dial”) in new stack
– Goto (main,3504,4)
– Executing [3504@main:4] AGI(“SIP/4999-0831e6d0”, “dial_setup.agi|3504”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dial_setup.agi
– AGI Script dial_setup.agi completed, returning 0
– Executing [3504@main:5] Dial(“SIP/4999-0831e6d0”, “SIP/3504|20|wW”) in new stack
Audio is at 10.0.0.1 port 13578
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
INVITE sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Contact: sip:4999@10.0.0.1
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:57:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 4793 4793 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 13578 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called 3504
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Allow-Events: talk,hold,conference
Content-Length: 0
<------------->
— (10 headers 0 lines) —
– SIP/3504-0831a410 is ringing
Scheduling destruction of SIP dialog ‘0fed059875d7fba14ea5b45214429272@10.0.0.1’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Scheduling destruction of SIP dialog ‘0fed059875d7fba14ea5b45214429272@10.0.0.1’ in 6400 ms (Method: INVITE)
== Spawn extension (main, 3504, 5) exited non-zero on 'SIP/4999-0831e6d0’
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 10.0.0.178:5060:
ACK sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
Contact: sip:4999@10.0.0.1
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Retransmitting #1 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #2 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #3 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #4 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178;tag=4F70555F-C5692AAA
CSeq: 102 INVITE
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
Contact: sip:3504@10.0.0.178
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #5 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Retransmitting #6 (no NAT) to 10.0.0.178:5060:
CANCEL sip:3504@10.0.0.178 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk*CLI> sip set debug off
<— SIP read from 10.0.0.178:5060 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK33cc0844;rport
From: “Charles Wonsey” sip:4999@10.0.0.1;tag=as7302b8a8
To: sip:3504@10.0.0.178
CSeq: 102 CANCEL
Call-ID: 0fed059875d7fba14ea5b45214429272@10.0.0.1
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Content-Length: 0
<------------->
— (8 headers 0 lines) — off
Calling to the broken extension - this is where I’m having the problem that the extension is not found
asteriskCLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
asteriskCLI> sip set debug peer 3504
SIP Debugging Enabled for IP: 10.0.0.178:5060
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
INVITE sip:4999@10.0.0.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone
CSeq: 1 INVITE
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 1400093487 1400093487 IN IP4 10.0.0.178
s=Polycom IP Phone
c=IN IP4 10.0.0.178
t=0 0
a=sendrecv
m=audio 2258 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (14 headers 11 lines) —
Sending to 10.0.0.178 : 5060 (no NAT)
Using INVITE request as basis request - bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
asterisk*CLI>
<— Reliably Transmitting (no NAT) to 10.0.0.178:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA;received=10.0.0.178
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7eb74102"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178’ in 32000 ms (Method: INVITE)
Found user '3504’
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
ACK sip:4999@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bK273c44d1E83C10CA
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
CSeq: 1 ACK
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Max-Forwards: 70
Content-Length: 0
<------------->
— (11 headers 0 lines) —
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
INVITE sip:4999@10.0.0.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone
CSeq: 2 INVITE
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username=“3504”, realm=“asterisk”, nonce=“7eb74102”, uri=“sip:4999@10.0.0.1:5060;user=phone”, response=“f6ae3c2a135d402882adda36b41b27f0”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 1400093487 1400093487 IN IP4 10.0.0.178
s=Polycom IP Phone
c=IN IP4 10.0.0.178
t=0 0
a=sendrecv
m=audio 2258 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
— (15 headers 11 lines) —
Sending to 10.0.0.178 : 5060 (no NAT)
Using INVITE request as basis request - bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Found user '3504’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.178:2258
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.178:2258
Looking for 4999 in main (domain 10.0.0.1)
asterisk*CLI>
<— Reliably Transmitting (no NAT) to 10.0.0.178:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929;received=10.0.0.178
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[May 14 13:51:32] NOTICE[4875]: chan_sip.c:13922 handle_request_invite: Call from ‘3504’ to extension ‘4999’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178’ in 32000 ms (Method: INVITE)
asterisk*CLI>
<— SIP read from 10.0.0.178:5060 —>
ACK sip:4999@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.178;branch=z9hG4bKcf6397c8386D929
From: “Alfreda Benford” sip:3504@10.0.0.1;tag=9009CFAB-EC311066
To: sip:4999@10.0.0.1;user=phone;tag=as37fea988
CSeq: 2 ACK
Call-ID: bacc6ff-2fc5f41d-e7c1fa00@10.0.0.178
Contact: sip:3504@10.0.0.178
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Proxy-Authorization: Digest username=“3504”, realm=“asterisk”, nonce=“7eb74102”, uri=“sip:4999@10.0.0.1:5060;user=phone”, response=“f6ae3c2a135d402882adda36b41b27f0”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
— (12 headers 0 lines) —
asterisk*CLI> sip set debug peer 4999
SIP Debugging Enabled for IP: 10.0.0.213:5060
Reliably Transmitting (no NAT) to 10.0.0.213:5060:
OPTIONS sip:4999@10.0.0.213 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK01a6e809;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as662d3c34
To: sip:4999@10.0.0.213
Contact: sip:asterisk@10.0.0.1
Call-ID: 4f53c004403b3bbf7839e5471c563c58@10.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 May 2014 18:51:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
asterisk*CLI>
<— SIP read from 10.0.0.213:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK01a6e809;rport
From: “asterisk” sip:asterisk@10.0.0.1;tag=as662d3c34
To: sip:4999@10.0.0.213;tag=7D8ABBEE-50B33D1D
CSeq: 102 OPTIONS
Call-ID: 4f53c004403b3bbf7839e5471c563c58@10.0.0.1
Contact: sip:4999@10.0.0.213
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,timer,replaces
Content-Length: 0
<------------->
— (14 headers 0 lines) —