Having the register string 4685XXXXXXX:[email protected]/4685XXXXXXX. Enabled sip debug and got this:
code has been modified to hide my phone numbers
[code:1]
<-- SIP read from 192.168.10.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK30d592bc;received=192.168.10.2;rport=5060
From: “Unknown” sip:[email protected];tag=as05b56d63
Call-ID: [email protected]
CSeq: 102 OPTIONS
To: sip:[email protected];tag=1108055643148664060
Supported: timer, replaces
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, NOTIFY, REFER, INFO
Server: RIX67GW2/exp-4-0-3-x (Apr 12 2007)
Content-Length: 0
— (10 headers 0 lines) —
Destroying call '[email protected]’
trixbox*CLI>
<-- SIP read from 192.168.10.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK0c905495;received=192.168.10.2;rport=5060
From: “Unknown” sip:[email protected];tag=as56f2b078
Call-ID: [email protected]
CSeq: 102 OPTIONS
To: sip:[email protected];tag=1108055643148664060
Supported: timer, replaces
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, NOTIFY, REFER, INFO
Server: RIX67GW2/exp-4-0-3-x (Apr 12 2007)
Content-Length: 0
— (10 headers 0 lines) —
Destroying call '[email protected]’
trixbox*CLI>
<-- SIP read from 62.80.200.53:5060:
INVITE sip:[email protected] SIP/2.0
Record-Route: sip:62.80.200.53;ftag=e9852fba0cf9dbdbe15b96b702f7de01;lr
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK6d5a.78fc3a264ef0f150dbe63a603fcb89bb.0
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2600e5797bc478001ced7e8d60384797;rport=5061
Max-Forwards: 16
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 200 INVITE
Contact: Anonymous sip:62.80.200.53:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 4111820133-525472220-3107127311-4157622960
h323-conf-id: 4111820133-525472220-3107127311-4157622960
Content-Length: 415
Content-Type: application/sdp
v=0
o=Sippy 137785388 0 IN IP4 62.80.200.53
s=SIP Call
t=0 0
m=audio 52478 RTP/AVP 18 4 98 99 2 8 0 3 101
c=IN IP4 62.80.200.53
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:98 G726-16/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
— (16 headers 18 lines) —
Using INVITE request as basis request - [email protected]
Sending to 62.80.200.53 : 5060 (non-NAT)
Found peer 'CellipOUT’
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 62.80.200.53:52478
Found description format G729
Found description format G723
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4685XXXXXXX in from-sip-external (domain 192.168.10.2)
list_route: hop: sip:62.80.200.53;ftag=e9852fba0cf9dbdbe15b96b702f7de01;lr
Transmitting (NAT) to 62.80.200.53:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK6d5a.78fc3a264ef0f150dbe63a603fcb89bb.0;received=62.80.200.53
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2600e5797bc478001ced7e8d60384797;rport=5061
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Length: 0
Transmitting (NAT) to 62.80.200.53:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK6d5a.78fc3a264ef0f150dbe63a603fcb89bb.0;received=62.80.200.53
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2600e5797bc478001ced7e8d60384797;rport=5061
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected];tag=as3fddf795
Call-ID: [email protected]
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Length: 0
We’re at 192.168.10.2 port 18720
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.80.200.53:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK6d5a.78fc3a264ef0f150dbe63a603fcb89bb.0;received=62.80.200.53
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2600e5797bc478001ced7e8d60384797;rport=5061
Record-Route: sip:62.80.200.53;ftag=e9852fba0cf9dbdbe15b96b702f7de01;lr
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected];tag=as3fddf795
Call-ID: [email protected]
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 3587 3587 IN IP4 192.168.10.2
s=session
c=IN IP4 192.168.10.2
t=0 0
m=audio 18720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
trixboxCLI>
<-- SIP read from 62.80.200.53:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK6d5a.f3cf9fd658a4f53802c58dd847ef6db9.0
Via: SIP/2.0/UDP 62.80.200.53:5061;rport=5061;branch=z9hG4bK59e75a9cf068acf0799d6d7164940e36
Max-Forwards: 16
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected];tag=as3fddf795
Call-ID: [email protected]
CSeq: 200 ACK
Expires: 300
User-Agent: Sippy
trixboxCLI>
— (10 headers 0 lines) —
trixbox*CLI>
<-- SIP read from 62.80.200.53:5060:
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK7d5a.751eca94558a8fbe39e1be734b5c460d.0
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2b102549574b42de0580b8c9c57b927e;rport=5061
Max-Forwards: 16
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected];tag=as3fddf795
Call-ID: [email protected]
CSeq: 201 BYE
Contact: Anonymous sip:62.80.200.53:5061
Expires: 300
User-Agent: Sippy
cisco-GUID: 4111820133-525472220-3107127311-4157622960
h323-conf-id: 4111820133-525472220-3107127311-4157622960
— (13 headers 0 lines) —
Sending to 62.80.200.53 : 5060 (NAT)
Transmitting (NAT) to 62.80.200.53:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.80.200.53;branch=z9hG4bK7d5a.751eca94558a8fbe39e1be734b5c460d.0;received=62.80.200.53
Via: SIP/2.0/UDP 62.80.200.53:5061;branch=z9hG4bK2b102549574b42de0580b8c9c57b927e;rport=5061
From: sip:[email protected];tag=e9852fba0cf9dbdbe15b96b702f7de01
To: sip:[email protected];tag=as3fddf795
Call-ID: [email protected]
CSeq: 201 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[email protected]
Content-Length: 0
Destroying call ‘[email protected]’
[/code:1]
A voice from my voip box says that the number can not be reached.
Please advice.