I have inbound routes that work fine when NAT=yes so I am pretty sure it is not the them. Here is the debug
With NAT=YES in the sip.conf (incoming will work fine but outbound fails with this setting)
[code]=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.05.03 14:38:01 =~=~=~=~=~=~=~=~=~=~=~=
sip ~Content-Length: 0Max-Forwards: 13CSeq: 101 ACKsip debug
asterisk1*CLI>
SIP Debugging enabled
asterisk1*CLI> si
Destroying call ‘3e214313799e649a5f852e3a2e1883f3@192.168.1.151’
asterisk1*CLI> sip
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.115.20.41:5060:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK1a5f0193;rport
From: sip:15125354351@sphone.vopr.vonage.net;tag=as1a1a1fd9
To: sip:15125354351@sphone.vopr.vonage.net
Call-ID: 3e214313799e649a5f852e3a2e1883f3@192.168.1.151
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354351”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“f2b6ca597cbbc68c86e2a89b726877af”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI> sip
Destroying call '23547b845fb4fd90248427363aa09859@192.168.1.151’
Destroying call ‘75bcc24111e8111375ba1b2f10710ede@192.168.1.151’
asterisk1*CLI> sip
<-- SIP read from 216.115.20.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK1a5f0193;rport
From: sip:15125354351@sphone.vopr.vonage.net;tag=as1a1a1fd9
To: sip:15125354351@sphone.vopr.vonage.net
Call-ID: 3e214313799e649a5f852e3a2e1883f3@192.168.1.151
CSeq: 125 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘3e214313799e649a5f852e3a2e1883f3@192.168.1.151’ in 32000 ms
asterisk1*CLI> sip
Destroying call ‘07be1d192399024f5adbc286514fbf54@192.168.1.151’
asterisk1*CLI> sip
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.115.20.41:5060:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK677ce886;rport
From: sip:15125354349@sphone.vopr.vonage.net;tag=as19dd3a31
To: sip:15125354349@sphone.vopr.vonage.net
Call-ID: 23547b845fb4fd90248427363aa09859@192.168.1.151
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354349”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“eb1ec17cd771a20b4ada8081a4462c48”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI> sip
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 216.115.20.41:5060:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK5ea6bbc9;rport
From: sip:15125354350@sphone.vopr.vonage.net;tag=as64f740c1
To: sip:15125354350@sphone.vopr.vonage.net
Call-ID: 75bcc24111e8111375ba1b2f10710ede@192.168.1.151
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354350”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“eeb1913f41840584d34c89cb74300e6e”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI> sip
REGISTER 13 headers, 0 lines
asterisk1*CLI> sip
Reliably Transmitting (NAT) to 216.115.20.41:5060:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK393c19cb;rport
From: sip:15125354348@sphone.vopr.vonage.net;tag=as6fb6483f
To: sip:15125354348@sphone.vopr.vonage.net
Call-ID: 07be1d192399024f5adbc286514fbf54@192.168.1.151
CSeq: 125 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354348”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“4680222f187de6eff1ebde295a6d4873”, opaque=""
Expires: 120
Contact: sip:113@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI> sip
<-- SIP read from 216.115.20.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK677ce886;rport
From: sip:15125354349@sphone.vopr.vonage.net;tag=as19dd3a31
To: sip:15125354349@sphone.vopr.vonage.net
Call-ID: 23547b845fb4fd90248427363aa09859@192.168.1.151
CSeq: 125 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘23547b845fb4fd90248427363aa09859@192.168.1.151’ in 32000 ms
asterisk1*CLI> sip
<-- SIP read from 216.115.20.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK393c19cb;rport
From: sip:15125354348@sphone.vopr.vonage.net;tag=as6fb6483f
To: sip:15125354348@sphone.vopr.vonage.net
Call-ID: 07be1d192399024f5adbc286514fbf54@192.168.1.151
CSeq: 125 REGISTER
Contact: sip:113@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘07be1d192399024f5adbc286514fbf54@192.168.1.151’ in 32000 ms
asterisk1*CLI> sip
<-- SIP read from 216.115.20.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK5ea6bbc9;rport
From: sip:15125354350@sphone.vopr.vonage.net;tag=as64f740c1
To: sip:15125354350@sphone.vopr.vonage.net
Call-ID: 75bcc24111e8111375ba1b2f10710ede@192.168.1.151
CSeq: 125 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘75bcc24111e8111375ba1b2f10710ede@192.168.1.151’ in 32000 ms
asterisk1*CLI> sip no
<-- SIP read from 216.115.20.41:5060:
INVITE sip:15125354348@68.201.250.45:5060 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.30.25:5060
Via: SIP/2.0/UDP 208.49.157.16:5060;branch=z9hG4bK19C7
Record-Route: sip:15125354348@216.115.20.41:5060
Record-Route: sip:15125354348@216.115.30.25:5060
From: “Cell Phone TX” sip:12817824574@208.49.157.16;tag=208625319
To: sip:15125354348@inbound2.vonage.net
Call-ID: 38BAA272-DA1311DA-950D8A76-B2E81983@208.49.157.16
CSeq: 101 INVITE
Contact: sip:12817824574@208.49.157.16:5060
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 361
v=0
o=CiscoSystemsSIP-GW-UserAgent 1759 5393 IN IP4 208.49.157.16
s=SIP Call
c=IN IP4 208.49.157.16
t=0 0
m=audio 21084 RTP/AVP 0 18 2 100 101
c=IN IP4 208.49.157.16
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
— (14 headers 15 lines)—
Using INVITE request as basis request - 38BAA272-DA1311DA-950D8A76-B2E81983@208.49.157.16
Sending to 216.115.20.41 : 5060 (NAT)
Found peer 'vonage_in_4351’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 208.49.157.16:21084
asterisk1*CLI> sip no
Found description format PCMU
Found description format G729
Found description format G726-32
Found description format X-NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x114 (ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 15125354348 in from-pstn (domain 68.201.250.45)
list_route: hop: sip:15125354348@216.115.20.41:5060
list_route: hop: sip:15125354348@216.115.30.25:5060
list_route: hop: sip:12817824574@208.49.157.16:5060
Transmitting (NAT) to 216.115.20.41:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.115.20.41:5060;received=216.115.20.41
Via: SIP/2.0/UDP 216.115.30.25:5060
Via: SIP/2.0/UDP 208.49.157.16:5060;branch=z9hG4bK19C7
From: “Cell Phone TX” sip:12817824574@208.49.157.16;tag=208625319
To: sip:15125354348@inbound2.vonage.net
Call-ID: 38BAA272-DA1311DA-950D8A76-B2E81983@208.49.157.16
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:15125354348@68.201.250.45
Content-Length: 0
asterisk1*CLI> sip no
– Executing Goto(“SIP/15125354351-3579”, “from-external-custom|attendant|1”) in new stack
asterisk1*CLI> sip no
– Goto (from-external-custom,attendant,1)
asterisk1*CLI> sip no
– Executing Answer(“SIP/15125354351-3579”, “”) in new stack
asterisk1*CLI> sip no
We’re at 68.201.250.45 port 12312
asterisk1*CLI> sip no
Adding codec 0x4 (ulaw) to SDP
asterisk1*CLI> sip no
Adding non-codec 0x1 (telephone-event) to SDP
asterisk1*CLI> sip no
Reliably Transmitting (NAT) to 216.115.20.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.20.41:5060;received=216.115.20.41
Via: SIP/2.0/UDP 216.115.30.25:5060
Via: SIP/2.0/UDP 208.49.157.16:5060;branch=z9hG4bK19C7
Record-Route: sip:15125354348@216.115.20.41:5060
Record-Route: sip:15125354348@216.115.30.25:5060
From: “Cell Phone TX” sip:12817824574@208.49.157.16;tag=208625319
To: sip:15125354348@inbound2.vonage.net;tag=as12d599a0
Call-ID: 38BAA272-DA1311DA-950D8A76-B2E81983@208.49.157.16
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:15125354348@68.201.250.45
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 2798 2798 IN IP4 68.201.250.45
s=session
c=IN IP4 68.201.250.45
t=0 0
m=audio 12312 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
asterisk1*CLI> sip no
– Executing Wait(“SIP/15125354351-3579”, “1”) in new stack
asterisk1*CLI> sip no
<-- SIP read from 216.115.20.41:5060:
ACK sip:15125354348@68.201.250.45:5060 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5060
Via: SIP/2.0/UDP 216.115.30.25:5060
Via: SIP/2.0/UDP 208.49.157.16:5060;branch=z9hG4bK7A6
From: “Cell Phone TX” sip:12817824574@208.49.157.16;tag=208625319
To: sip:15125354348@inbound2.vonage.net;tag=as12d599a0
Call-ID: 38BAA272-DA1311DA-950D8A76-B2E81983@208.49.157.16
CSeq: 101 ACK
Max-Forwards: 13
Content-Length: 0
— (10 headers 0 lines)—
asterisk1*CLI> sip no debu
– Executing BackGround(“SIP/15125354351-3579”, “pls-wait-connect-call”) in new stack
– Playing ‘pls-wait-connect-call’ (language ‘en’)
asterisk1*CLI> sip no debug
– Executing Dial(“SIP/15125354351-3579”, “local/1@from-internal|30|mt”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing VoiceMail(“SIP/15125354351-3579”, “200@default”) in new stack
asterisk1*CLI> sip no debug
asterisk1*CLI>
SIP Debugging Disabled
asterisk1*CLI> [/code]
Here is the debug with NAT=NO in the sip.conf no ther changes (outbund will now work but inbound dies)
[code]
asterisk1*CLI>
Destroying call ‘3e214313799e649a5f852e3a2e1883f3@192.168.1.151’
asterisk1*CLI>
Destroying call ‘23547b845fb4fd90248427363aa09859@192.168.1.151’
asterisk1*CLI>
Destroying call ‘07be1d192399024f5adbc286514fbf54@192.168.1.151’
asterisk1*CLI>
Destroying call ‘75bcc24111e8111375ba1b2f10710ede@192.168.1.151’
asterisk1*CLI>
<-- SIP read from 216.115.20.41:5061:
INVITE sip:15125354348@68.201.250.45:5060 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5061
Via: SIP/2.0/UDP 216.115.20.24:5060
Via: SIP/2.0/UDP 208.49.157.17:5060;branch=z9hG4bK90A
Record-Route: sip:15125354348@216.115.20.41:5061
Record-Route: sip:15125354348@216.115.20.24:5060
From: “Cell Phone TX” sip:12817824574@208.49.157.17;tag=1894722780
To: sip:15125354348@inbound4.vonage.net
Call-ID: DCC88D50-DA1311DA-BDC9ECAF-6A62EE8@208.49.157.17
CSeq: 101 INVITE
Contact: sip:12817824574@208.49.157.17:5060
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 361
v=0
o=CiscoSystemsSIP-GW-UserAgent 2414 7130 IN IP4 208.49.157.17
s=SIP Call
c=IN IP4 208.49.157.17
t=0 0
m=audio 19884 RTP/AVP 0 18 2 100 101
c=IN IP4 208.49.157.17
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
— (14 headers 15 lines)—
Using INVITE request as basis request - DCC88D50-DA1311DA-BDC9ECAF-6A62EE8@208.49.157.17
Sending to 216.115.20.41 : 5061 (non-NAT)
Found peer ‘vonage_out_4351’
asterisk1*CLI>
Reliably Transmitting (NAT) to 216.115.20.41:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 216.115.20.41:5061;received=216.115.20.41
Via: SIP/2.0/UDP 216.115.20.24:5060
Via: SIP/2.0/UDP 208.49.157.17:5060;branch=z9hG4bK90A
From: “Cell Phone TX” sip:12817824574@208.49.157.17;tag=1894722780
To: sip:15125354348@inbound4.vonage.net;tag=as7daab599
Call-ID: DCC88D50-DA1311DA-BDC9ECAF-6A62EE8@208.49.157.17
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:15125354348@68.201.250.45
Proxy-Authenticate: Digest realm=“asterisk”, nonce=“17e2378d”
Content-Length: 0
Scheduling destruction of call ‘DCC88D50-DA1311DA-BDC9ECAF-6A62EE8@208.49.157.17’ in 15000 ms
asterisk1*CLI>
<-- SIP read from 216.115.20.41:5061:
ACK sip:15125354348@inbound4.vonage.net SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5061
From: “Cell Phone TX” sip:12817824574@208.49.157.17;tag=1894722780
To: sip:15125354348@inbound4.vonage.net;tag=as7daab599
Call-ID: DCC88D50-DA1311DA-BDC9ECAF-6A62EE8@208.49.157.17
CSeq: 101 ACK
Content-Length: 0
— (7 headers 0 lines)—
asterisk1*CLI>
Destroying call ‘61e5ff4e71d6e2261c0b02e5111f4681@192.168.1.151’
asterisk1*CLI>
Destroying call ‘3152a9505b64ba8e2a9e2e3c63b44ac4@192.168.1.151’
asterisk1*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK1f8ffa9a;rport
From: sip:15125354351@sphone.vopr.vonage.net;tag=as3a2eb01e
To: sip:15125354351@sphone.vopr.vonage.net
Call-ID: 61e5ff4e71d6e2261c0b02e5111f4681@192.168.1.151
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354351”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“f2b6ca597cbbc68c86e2a89b726877af”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI>
Destroying call ‘462a6ada02df37c41ac7e759558124f8@192.168.1.151’
asterisk1*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK6eeab06c;rport
From: sip:15125354350@sphone.vopr.vonage.net;tag=as2af300bb
To: sip:15125354350@sphone.vopr.vonage.net
Call-ID: 3152a9505b64ba8e2a9e2e3c63b44ac4@192.168.1.151
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354350”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“eeb1913f41840584d34c89cb74300e6e”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI>
<-- SIP read from 216.115.20.41:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK1f8ffa9a;rport
From: sip:15125354351@sphone.vopr.vonage.net;tag=as3a2eb01e
To: sip:15125354351@sphone.vopr.vonage.net
Call-ID: 61e5ff4e71d6e2261c0b02e5111f4681@192.168.1.151
CSeq: 105 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘61e5ff4e71d6e2261c0b02e5111f4681@192.168.1.151’ in 32000 ms
asterisk1*CLI>
Destroying call ‘558d44767068f4a752d3b9be3cacffe5@192.168.1.151’
asterisk1*CLI>
<-- SIP read from 216.115.20.41:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK6eeab06c;rport
From: sip:15125354350@sphone.vopr.vonage.net;tag=as2af300bb
To: sip:15125354350@sphone.vopr.vonage.net
Call-ID: 3152a9505b64ba8e2a9e2e3c63b44ac4@192.168.1.151
CSeq: 105 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘3152a9505b64ba8e2a9e2e3c63b44ac4@192.168.1.151’ in 32000 ms
asterisk1*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK72aae41e;rport
From: sip:15125354349@sphone.vopr.vonage.net;tag=as047dead6
To: sip:15125354349@sphone.vopr.vonage.net
Call-ID: 462a6ada02df37c41ac7e759558124f8@192.168.1.151
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354349”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“eb1ec17cd771a20b4ada8081a4462c48”, opaque=""
Expires: 120
Contact: sip:s@68.201.250.45
Event: registration
Content-Length: 0
asterisk1*CLI>
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK3123b759;rport
From: sip:15125354348@sphone.vopr.vonage.net;tag=as12280c2d
To: sip:15125354348@sphone.vopr.vonage.net
Call-ID: 558d44767068f4a752d3b9be3cacffe5@192.168.1.151
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“15125354348”, realm=“216.115.20.41”, algorithm=MD5, uri=“sip:216.115.20.41”, nonce=“1534782874”, response=“4680222f187de6eff1ebde295a6d4873”, opaque=""
Expires: 120
Contact: sip:113@68.201.250.45
Event: registration
Content-Length: 0
<-- SIP read from 216.115.20.41:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK72aae41e;rport
From: sip:15125354349@sphone.vopr.vonage.net;tag=as047dead6
To: sip:15125354349@sphone.vopr.vonage.net
Call-ID: 462a6ada02df37c41ac7e759558124f8@192.168.1.151
CSeq: 105 REGISTER
Contact: sip:s@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘462a6ada02df37c41ac7e759558124f8@192.168.1.151’ in 32000 ms
asterisk1*CLI>
<-- SIP read from 216.115.20.41:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.201.250.45:5060;branch=z9hG4bK3123b759;rport
From: sip:15125354348@sphone.vopr.vonage.net;tag=as12280c2d
To: sip:15125354348@sphone.vopr.vonage.net
Call-ID: 558d44767068f4a752d3b9be3cacffe5@192.168.1.151
CSeq: 105 REGISTER
Contact: sip:113@68.201.250.45;expires=20
Max-Forwards: 70
Content-Length: 0
— (9 headers 0 lines)—
Scheduling destruction of call ‘558d44767068f4a752d3b9be3cacffe5@192.168.1.151’ in 32000 ms
asterisk1*CLI> sip debugno debug[/code]
Thanks so much for the response, if you guys can figure out what is going on I would be greatful.
Thanks,
Nick