Recently I decided to set up Asterisk on my main server (Gentoo x64) rather than continue with an independent server running trixbox (Asterisk 1.2.13). This means i currently have a working trixbox install and a non-working from-source install to compare to each other. During the troubleshooting below, both servers were not running asterisk at the same time (/etc/init.d/asterisk stop on one before starting the other)
First I tried 1.6 beta, now am using 1.4.17, as I thought some synax change or beta issue was the problem. It is apparently not. I built both from source, and uninstalled 1.6 before installing 1.4.17. There don’t appear to me any errors with the build or in the starup of Asterisk.
Here is the log from the last asterisk startup:
[Jan 20 23:30:52] NOTICE[8463] cdr.c: CDR simple logging enabled.
[Jan 20 23:30:52] NOTICE[8463] loader.c: 146 modules will be loaded.
[Jan 20 23:30:52] WARNING[8463] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: Starting AEL load process.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:52] NOTICE[8463] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Jan 20 23:30:53] NOTICE[8477] chan_sip.c: Peer 'ViatalkSIP' is now Reachable. (75ms / 2000ms)
I am attempting to register with Viatalk via SIP, and have that channel passed to the context listed in the peer definition.
The context was built using the O’reilly simple context of Answer, Playback, Hangup.
Context Snippet from extensions.conf:
[from-sip-trunk]
exten => s,1,Answer()
exten => s,n,Playback(demo-congrats.gsm)
exten => s,n,Hangup()
; end of [from-sip-trunk]
Asterisk is aware of the context:
BigBox*CLI> dialplan show from-sip-trunk
[ Context 'from-sip-trunk' created by 'pbx_config' ]
's' => 1. Answer() [pbx_config]
2. Playback(demo-congrats.gsm) [pbx_config]
3. Hangup() [pbx_config]
Peer snippet from sip.conf:
register=1541705xxxx:xxxxxxxxxx@sanfrancisco-1.vtnoc.net
[ViatalkSIP]
username=1541705xxxx
type=peer
secret=xxxxxxxxx
qualify=yes
insecure=port,invite
host=sanfrancisco-1.vtnoc.net
fromuser=1541705xxxx
fromdomain=sanfrancisco-1.vtnoc.net
dtmfmode=inband
dtmf=auto
disallow=all
context=from-sip-trunk
canreinvite=yes
authuser=1541705xxxx
allow=ulaw
Registration never seems to happen. Setting the debug flag shows every Options (102) packet sent receives a 404 from Viatalk (if i’m reading this correctly).
BigBox*CLI> sip set debug peer ViatalkSIP
SIP Debugging Enabled for IP: 216.246.105.146:5060
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
OPTIONS sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.10:5060;branch=z9hG4bK1712b2a9;rport
From: "asterisk" <sip:asterisk@192.168.11.10>;tag=as208fc522
To: <sip:sanfrancisco-1.vtnoc.net>
Contact: <sip:asterisk@192.168.11.10>
Call-ID: 6d8eaf1552140c411d852c5c681181b3@192.168.11.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Jan 2008 07:45:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
BigBox*CLI>
<--- SIP read from 216.246.105.146:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 71.94.236.209:5060;branch=z9hG4bK1712b2a9;received=71.94.236.209;rport=5060
From: "asterisk" <sip:asterisk@192.168.11.10>;tag=as208fc522
To: <sip:sanfrancisco-1.vtnoc.net>;tag=as447af17e
Call-ID: 6d8eaf1552140c411d852c5c681181b3@192.168.11.10
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
BigBox*CLI>
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '6d8eaf1552140c411d852c5c681181b3@192.168.11.10' Method: OPTIONS
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
OPTIONS sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.10:5060;branch=z9hG4bK70a08aa1;rport
From: "asterisk" <sip:asterisk@192.168.11.10>;tag=as3c61deca
To: <sip:sanfrancisco-1.vtnoc.net>
Contact: <sip:asterisk@192.168.11.10>
Call-ID: 7025f648535b3aa2155c577d469476bb@192.168.11.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 21 Jan 2008 07:46:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
BigBox*CLI>
<--- SIP read from 216.246.105.146:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 71.94.236.209:5060;branch=z9hG4bK70a08aa1;received=71.94.236.209;rport=5060
From: "asterisk" <sip:asterisk@192.168.11.10>;tag=as3c61deca
To: <sip:sanfrancisco-1.vtnoc.net>;tag=as5250793a
Call-ID: 7025f648535b3aa2155c577d469476bb@192.168.11.10
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
BigBox*CLI>
The trixbox install is still working, even though I see some 404’s and 401s from Viatalk.
asterisk1*CLI> sip show registry
Host Username Refresh State
sanfrancisco-1.vtnoc.net:5060 1541705xxxx 105 Registered
Here is a snippet from the working trixbox sip.conf:
register=1541705xxxx:xxxxxxxxxx@sanfrancisco-1.vtnoc.net
[ViatalkSIP]
username=1541705xxxx
type=peer
secret=xxxxxxxxxx
qualify=yes
insecure=very
host=sanfrancisco-1.vtnoc.net
fromuser=1541705xxxx
fromdomain=sanfrancisco-1.vtnoc.net
dtmfmode=inband
dtmf=auto
disallow=all
context=from-trunk
canreinvite=yes
authuser=1541705xxxx
allow=ulaw
and here is the debug output from the working server asterisk -r, showing 404s and 401s:
asterisk1*CLI> sip debug peer ViatalkSIP
SIP Debugging Enabled for IP: 216.246.105.146:5060
Destroying call '5edf81661626fc6366f5e884037899a7@127.0.0.1'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
OPTIONS sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK5dff7406;rport
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as5b031151
To: <sip:sanfrancisco-1.vtnoc.net>
Contact: <sip:Unknown@192.168.11.11>
Call-ID: 5fd41fcc5caa6cfa4e6e403669784c5a@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:43:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK5dff7406;received=71.94.236.209;rport=1024
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as5b031151
To: <sip:sanfrancisco-1.vtnoc.net>;tag=as37c70ca4
Call-ID: 5fd41fcc5caa6cfa4e6e403669784c5a@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '5fd41fcc5caa6cfa4e6e403669784c5a@192.168.11.11'
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
OPTIONS sip:s@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK5a543db8;rport
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as6efa984b
To: <sip:s@192.168.11.11>
Contact: <sip:asterisk@216.246.105.146>
Call-ID: 50e0794954db24943ead03174be27082@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:43:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.11.11)
Transmitting (no NAT) to 216.246.105.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK5a543db8;received=216.246.105.146;rport=5060
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as6efa984b
To: <sip:s@192.168.11.11>;tag=as50ce0239
Call-ID: 50e0794954db24943ead03174be27082@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.11.11>
Accept: application/sdp
Content-Length: 0
---
Destroying call '50e0794954db24943ead03174be27082@216.246.105.146'
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
REGISTER sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK7aee35e7;rport
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as526557dc
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1541705xxxx", realm="asterisk", algorithm=MD5, uri="sip:sanfrancisco-1.vtnoc.net", nonce="4c73f20b", response="1fcff740556e719624760a1f423a6035", opaque=""
Expires: 120
Contact: <sip:s@192.168.11.11>
Event: registration
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK7aee35e7;received=71.94.236.209;rport=1024
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as526557dc
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1541705xxxx@216.246.105.146>
Content-Length: 0
--- (11 headers 0 lines) ---
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK7aee35e7;received=71.94.236.209;rport=1024
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as526557dc
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as31e36174
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="075a96d5"
Content-Length: 0
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name sanfrancisco-1.vtnoc.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
REGISTER sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK079aa3b4;rport
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as0f869aef
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1541705xxxx", realm="asterisk", algorithm=MD5, uri="sip:sanfrancisco-1.vtnoc.net", nonce="075a96d5", response="c350046d77eb492446b32be6d0ff6c39", opaque=""
Expires: 120
Contact: <sip:s@192.168.11.11>
Event: registration
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK079aa3b4;received=71.94.236.209;rport=1024
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as0f869aef
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1541705xxxx@216.246.105.146>
Content-Length: 0
--- (11 headers 0 lines) ---
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK079aa3b4;received=71.94.236.209;rport=1024
From: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as0f869aef
To: <sip:1541705xxxx@sanfrancisco-1.vtnoc.net>;tag=as31e36174
Call-ID: 5edf81661626fc6366f5e884037899a7@127.0.0.1
CSeq: 105 REGISTER
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:s@192.168.11.11>;expires=120
Date: Sun, 20 Jan 2008 23:43:46 GMT
Content-Length: 0
--- (13 headers 0 lines) ---
Scheduling destruction of call '5edf81661626fc6366f5e884037899a7@127.0.0.1' in 32000 ms
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
NOTIFY sip:s@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK566b7f61;rport
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as7b332eec
To: <sip:s@192.168.11.11>
Contact: <sip:asterisk@216.246.105.146>
Call-ID: 11d038a049155cbe0367e2956f92302b@216.246.105.146
CSeq: 102 NOTIFY
User-Agent: Viatalk SIP
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@216.246.105.146
Voice-Message: 1/0 (0/0)
--- (12 headers 3 lines) ---
Transmitting (no NAT) to 216.246.105.146:5060:
SIP/2.0 603 Declined (no dialog)
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK566b7f61;received=216.246.105.146;rport=5060
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as7b332eec
To: <sip:s@192.168.11.11>;tag=as5eb1046b
Call-ID: 11d038a049155cbe0367e2956f92302b@216.246.105.146
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
OPTIONS sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK46fef114;rport
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as3d6c0ffb
To: <sip:sanfrancisco-1.vtnoc.net>
Contact: <sip:Unknown@192.168.11.11>
Call-ID: 708f420c64bc32d7321d4bb4453baf12@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:44:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK46fef114;received=71.94.236.209;rport=1024
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as3d6c0ffb
To: <sip:sanfrancisco-1.vtnoc.net>;tag=as64a455f4
Call-ID: 708f420c64bc32d7321d4bb4453baf12@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '708f420c64bc32d7321d4bb4453baf12@192.168.11.11'
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
OPTIONS sip:s@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK1f389ccf;rport
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as1483ccf1
To: <sip:s@192.168.11.11>
Contact: <sip:asterisk@216.246.105.146>
Call-ID: 60713b3114d86b592c7cdcc312ade610@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:44:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.11.11)
Transmitting (no NAT) to 216.246.105.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK1f389ccf;received=216.246.105.146;rport=5060
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as1483ccf1
To: <sip:s@192.168.11.11>;tag=as146a24c3
Call-ID: 60713b3114d86b592c7cdcc312ade610@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.11.11>
Accept: application/sdp
Content-Length: 0
---
Destroying call '60713b3114d86b592c7cdcc312ade610@216.246.105.146'
Destroying call '5edf81661626fc6366f5e884037899a7@127.0.0.1'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 216.246.105.146:5060:
OPTIONS sip:sanfrancisco-1.vtnoc.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK0adea207;rport
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as3e7d4ddd
To: <sip:sanfrancisco-1.vtnoc.net>
Contact: <sip:Unknown@192.168.11.11>
Call-ID: 54f3cb392ff6dc8a526d24cf2edbc1ed@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:45:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.11.11:5060;branch=z9hG4bK0adea207;received=71.94.236.209;rport=1024
From: "Unknown" <sip:Unknown@192.168.11.11>;tag=as3e7d4ddd
To: <sip:sanfrancisco-1.vtnoc.net>;tag=as73a23222
Call-ID: 54f3cb392ff6dc8a526d24cf2edbc1ed@192.168.11.11
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '54f3cb392ff6dc8a526d24cf2edbc1ed@192.168.11.11'
asterisk1*CLI>
<-- SIP read from 216.246.105.146:5060:
OPTIONS sip:s@192.168.11.11 SIP/2.0
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK12e3c53d;rport
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as209e0ae5
To: <sip:s@192.168.11.11>
Contact: <sip:asterisk@216.246.105.146>
Call-ID: 73ceff352d40ed9f23d24b8c0e26b4ca@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Viatalk SIP
Max-Forwards: 70
Date: Sun, 20 Jan 2008 23:45:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.11.11)
Transmitting (no NAT) to 216.246.105.146:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.246.105.146:5060;branch=z9hG4bK12e3c53d;received=216.246.105.146;rport=5060
From: "asterisk" <sip:asterisk@216.246.105.146>;tag=as209e0ae5
To: <sip:s@192.168.11.11>;tag=as6fb205f4
Call-ID: 73ceff352d40ed9f23d24b8c0e26b4ca@216.246.105.146
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.11.11>
Accept: application/sdp
Content-Length: 0
Additionally, I have packet captures of both if they are needed for diagnosing.
What am I missing here guys? I’ve been pouring over it for hours…
Thanks!
thenextdon13