Hi John
Thanks for your reply.
I see that trunk had some issues and now the sip show peers shows registered
Host dnsmgr Username Refresh State Reg.Time
xx.xxx.x.xx:5060 N username 45 Registered Sat, 11 May 2013 18:43:16
1 SIP registrations.
I add this to this 0ZXXXXXXXX to the new outbound route
0 = prepend+prefix =Z|match pattern is XXXXXXXX
this is how the GUI has it.
I still cant get a call out. here is sip trace: (i just noticed that there is not trace of this sip information going to the ISP my is an ip starting with 41.xx.xx.xx)
[[email protected] ~]# asterisk -r
Asterisk 11.3.0, Copyright © 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
Connected to Asterisk 11.3.0 currently running on localhost (pid = 1614)
<— SIP read from UDP:192.168.10.202:6654 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:6654
To: sip:[email protected]
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 307
v=0
o=- 13012764492936112 1 IN IP4 192.168.10.202
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 192.168.10.202
t=0 0
m=audio 55800 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 192.168.10.202:6654 (no NAT)
Using INVITE request as basis request - ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
Found peer ‘1000’ for ‘1000’ from 192.168.10.202:6654
<— Reliably Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as7e28b2e8
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="278cced4"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.10.202:6654 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as7e28b2e8
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:192.168.10.202:6654 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:6654
To: sip:[email protected]
From: “Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.2 stamp 70142
Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“278cced4”,uri="sip:[email protected]”,response=“a881ff03b1ed49d095e3792149b11a5c”,algorithm=MD5
Content-Length: 307
v=0
o=- 13012764492936112 1 IN IP4 192.168.10.202
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 192.168.10.202
t=0 0
m=audio 55800 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 192.168.10.202:6654 (no NAT)
Using INVITE request as basis request - ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
Found peer ‘1000’ for ‘1000’ from 192.168.10.202:6654
Found RTP audio format 125
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found unknown media description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|speex16|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.202:55800
Looking for 00832898786 in from-internal (domain 192.168.10.16)
list_route: hop: sip:[email protected]:6654
<— Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected]
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
Audio is at 11334
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as45c2855e
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 985877677 985877677 IN IP4 192.168.10.16
s=Asterisk PBX 11.3.0
c=IN IP4 192.168.10.16
t=0 0
m=audio 11334 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=ptime:20
a=sendrecv
<------------>
Really destroying SIP dialog ‘[email protected][::1]’ Method: REGISTER
<— SIP read from UDP:192.168.10.202:6654 —>
<------------->
<— Reliably Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as45c2855e
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2013-05-11 18:48:23] WARNING[5914][C-00000034]: channel.c:4816 ast_prod: Prodding channel ‘SIP/1000-00000042’ failed
<— SIP read from UDP:192.168.10.202:6654 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as45c2855e
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU’ Method: ACK
Regards