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Thanks for your help. I've updated the wiki page at
<a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org/wiki/Provider_Configuration:_Link2VoIP#Link2VoIP">http://wiki.freeswitch.org/wiki/Provider_Configuration:_Link2VoIP#Link2VoIP</a><br>
<br>
Darren Wiebe<br>
<br>
On 10/01/2011 6:12 PM, Michael Collins wrote:
<blockquote
cite="mid:AANLkTin4348ir9zSAd=5PaqLjvzCexEATFeviHRsQA0j@mail.gmail.com"
type="cite">How many different DIDs do you have for this user?
Just one? If so can you not map the user to a specific DID? In any
case, throw a "info" app in your public dialplan and call the DID.
You'll see there's all sorts of variables you can use for routing
if you need to.
<div>
<br>
</div>
<div>-MC<br>
<br>
<div class="gmail_quote">On Mon, Jan 10, 2011 at 11:30 PM,
Darren Wiebe <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:darren@aleph-com.net">darren@aleph-com.net</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#ffffff"> Yeah, I bet. :) The
outgoing call problem was my fault, I had an incorrect
piece of dialplan. Here's the trace on an incoming call.
I'm trying to get it to come to a particular DID in the
public context instead of this.<br>
<br>
2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
Processing CLID NAME<br>
<7806283672>->sipuser in context public<br>
<br>
What am I missing?<br>
<br>
Here's the relevant provider entry from the external sip
profile<br>
<br>
<include><br>
<gateway name="link2voip1"><br>
<param name="username" value="sipuser"/><br>
<param name="password" value="password"/><br>
<param name="proxy" value="<a moz-do-not-send="true"
href="http://sip.ca1.link2voip.com" target="_blank">sip.ca1.link2voip.com</a>"/><br>
<param name="register" value="true"/><br>
<param name="register-transport" value="udp"/><br>
<param name="retry-seconds" value="30"/><br>
</gateway><br>
</include><br>
<br>
Sip Trace: <br>
<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a>;tag=as01e3f5de<br>
Call-ID: 0472a7de73e00fc578b12f6679b6dd9d@CUSTOMERIP<br>
To: <a moz-do-not-send="true"><sip:14034883602@sip.ca2.link2voip.com></a>;tag=m0Ur3NvDZma5H<br>
CSeq: 103 ACK<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
recv 942 bytes from udp/[66.51.110.210]:5060 at
23:22:36.415027:<br>
------------------------------------------------------------------------<br>
INVITE <a moz-do-not-send="true">sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2</a>
SIP/2.<br>
0<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.110.210;lr;ftag=as368d01dd></a><br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0<br>
Via: SIP/2.0/UDP
CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a>;tag=as368d01dd<br>
To: <a moz-do-not-send="true"><sip:14034883602@sip.ca2.link2voip.com></a><br>
Contact: <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a><br>
Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP<br>
CSeq: 103 INVITE<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 69<br>
Date: Mon, 10 Jan 2011 23:22:36 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY<br>
Content-Type: application/sdp<br>
Content-Length: 235<br>
<br>
v=0<br>
o=root 12790 12791 IN IP4 CUSTOMERIP<br>
s=session<br>
c=IN IP4 66.51.110.210<br>
t=0 0<br>
m=audio 14648 RTP/AVP 0 96<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:96 telephone-event/8000<br>
a=fmtp:96 0-16<br>
a=silenceSupp:off - - - -<br>
a=nortpproxy:yes<br>
------------------------------------------------------------------------<br>
send 503 bytes to udp/[66.51.110.210]:5060 at
23:22:36.415027:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0<br>
Via: SIP/2.0/UDP
CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.110.210;lr;ftag=as368d01dd></a><br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a>;tag=as368d01dd<br>
To: <a moz-do-not-send="true"><sip:14034883602@sip.ca2.link2voip.com></a><br>
Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP<br>
CSeq: 103 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784
New Channel sofia/exter<br>
nal/7806283672@CUSTOMERIP
[c58ce64a-c99e-4a20-94a9-680e488f8fab]<br>
2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331
Processing CLID NAME<br>
<7806283672>->sipuser in context public<br>
2011-01-10 16:22:36.430652 [NOTICE]
switch_core_state_machine.c:189 sofia/extern<br>
al/7806283672@CUSTOMERIP has executed the last dialplan
instruction, hangin<br>
g up.<br>
2011-01-10 16:22:36.430652 [NOTICE]
switch_core_state_machine.c:191 Hangup sofia<br>
/external/7806283672@CUSTOMERIP [CS_EXECUTE]
[NORMAL_CLEARING]<br>
send 818 bytes to udp/[66.51.110.210]:5060 at
23:22:36.430652:<br>
------------------------------------------------------------------------<br>
SIP/2.0 480 Temporarily Unavailable<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0<br>
Via: SIP/2.0/UDP
CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a>;tag=as368d01dd<br>
To: <a moz-do-not-send="true"><sip:14034883602@sip.ca2.link2voip.com></a>;tag=N9mH5gDHvX0QD<br>
Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP<br>
CSeq: 103 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
UPDATE, INFO, REGISTER, RE<br>
FER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, refer<br>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>
Content-Length: 0<br>
Remote-Party-ID: "sipuser" <a moz-do-not-send="true"><sip:sipuser@192.168.35.1></a>;party=calling;privacy=o<br>
ff;screen=no<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:36.430652 [NOTICE]
switch_core_session.c:1273 Session 53 (sofia<br>
/external/7806283672@CUSTOMERIP) Ended<br>
2011-01-10 16:22:36.430652 [NOTICE]
switch_core_session.c:1275 Close Channel sof<br>
ia/external/7806283672@CUSTOMERIP [CS_DESTROY]<br>
recv 367 bytes from udp/[66.51.110.210]:5060 at
23:22:36.696268:<br>
------------------------------------------------------------------------<br>
ACK <a moz-do-not-send="true">sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2</a>
SIP/2.0<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@CUSTOMERIP></a>;tag=as368d01dd<br>
Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP<br>
To: <a moz-do-not-send="true"><sip:14034883602@sip.ca2.link2voip.com></a>;tag=N9mH5gDHvX0QD<br>
CSeq: 103 ACK<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
recv 1230 bytes from udp/[66.51.110.210]:5060 at
23:22:37.789983:<br>
------------------------------------------------------------------------<br>
INVITE <a moz-do-not-send="true">sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2</a>
SIP/2.<br>
0<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e></a><br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F<br>
Max-Forwards: 66<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=ZNcB4yyH3SD3e<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.110.210></a><br>
Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
Contact: <a moz-do-not-send="true"><sip:ciscosip@66.51.127.163:5080></a><br>
User-Agent: Cisco-SIPGateway/IOS-12.x<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY,<br>
REFER, UPDATE, REGISTER, INFO<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 323<br>
Remote-Party-ID: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;screen=yes;pri<br>
vacy=off<br>
<br>
v=0<br>
o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072
338127983379325658 IN IP4<br>
66.51.127.163<br>
s=SIP Call<br>
c=IN IP4 66.51.110.210<br>
t=0 0<br>
m=audio 14650 RTP/AVP 0 18 101 13<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:18 G729/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
a=nortpproxy:yes<br>
------------------------------------------------------------------------<br>
send 494 bytes to udp/[66.51.110.210]:5060 at
23:22:37.789983:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e></a><br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=ZNcB4yyH3SD3e<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.110.210></a><br>
Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784
New Channel sofia/exter<br>
<a moz-do-not-send="true"
href="mailto:nal/7806283672@66.51.127.163"
target="_blank">nal/7806283672@66.51.127.163</a>
[86b78ecd-469f-4a1c-9fe5-692a5941ff37]<br>
2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331
Processing CLID NAME<br>
<7806283672>->sipuser in context public<br>
2011-01-10 16:22:37.805608 [NOTICE]
switch_core_state_machine.c:189 sofia/extern<br>
<a moz-do-not-send="true"
href="mailto:al/7806283672@66.51.127.163"
target="_blank">al/7806283672@66.51.127.163</a> has
executed the last dialplan instruction, hanging<br>
up.<br>
2011-01-10 16:22:37.805608 [NOTICE]
switch_core_state_machine.c:191 Hangup sofia<br>
/external/<a moz-do-not-send="true"
href="mailto:7806283672@66.51.127.163" target="_blank">7806283672@66.51.127.163</a>
[CS_EXECUTE] [NORMAL_CLEARING]<br>
send 806 bytes to udp/[66.51.110.210]:5060 at
23:22:37.805608:<br>
------------------------------------------------------------------------<br>
SIP/2.0 480 Temporarily Unavailable<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=ZNcB4yyH3SD3e<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.110.210></a>;tag=pjea7BymS6paS<br>
Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
UPDATE, INFO, REGISTER, RE<br>
FER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, refer<br>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>
Content-Length: 0<br>
Remote-Party-ID: "sipuser" <a moz-do-not-send="true"><sip:sipuser@192.168.35.1></a>;party=calling;privacy=o<br>
ff;screen=no<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:37.805608 [NOTICE]
switch_core_session.c:1273 Session 54 (sofia<br>
/external/<a moz-do-not-send="true"
href="mailto:7806283672@66.51.127.163" target="_blank">7806283672@66.51.127.163</a>)
Ended<br>
2011-01-10 16:22:37.805608 [NOTICE]
switch_core_session.c:1275 Close Channel sof<br>
<a moz-do-not-send="true"
href="mailto:ia/external/7806283672@66.51.127.163"
target="_blank">ia/external/7806283672@66.51.127.163</a>
[CS_DESTROY]<br>
recv 352 bytes from udp/[66.51.110.210]:5060 at
23:22:37.914979:<br>
------------------------------------------------------------------------<br>
ACK <a moz-do-not-send="true">sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2</a>
SIP/2.0<br>
Via: SIP/2.0/UDP
66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=ZNcB4yyH3SD3e<br>
Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.110.210></a>;tag=pjea7BymS6paS<br>
CSeq: 7014814 ACK<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
recv 1230 bytes from udp/[66.51.127.173]:5060 at
23:22:37.977477:<br>
------------------------------------------------------------------------<br>
INVITE <a moz-do-not-send="true">sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1</a>
SIP/2.<br>
0<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N></a><br>
Via: SIP/2.0/UDP
66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ<br>
Max-Forwards: 66<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=17yv7m0rXBt8N<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.127.173></a><br>
Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
Contact: <a moz-do-not-send="true"><sip:ciscosip@66.51.127.163:5080></a><br>
User-Agent: Cisco-SIPGateway/IOS-12.x<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY,<br>
REFER, UPDATE, REGISTER, INFO<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 323<br>
Remote-Party-ID: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;screen=yes;pri<br>
vacy=off<br>
<br>
v=0<br>
o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778
787843793424096957 IN IP4<br>
66.51.127.163<br>
s=SIP Call<br>
c=IN IP4 66.51.127.173<br>
t=0 0<br>
m=audio 15488 RTP/AVP 0 18 101 13<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:18 G729/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=rtpmap:13 CN/8000<br>
a=ptime:20<br>
a=nortpproxy:yes<br>
------------------------------------------------------------------------<br>
send 494 bytes to udp/[66.51.127.173]:5060 at
23:22:37.977477:<br>
------------------------------------------------------------------------<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP
66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ<br>
Record-Route: <a moz-do-not-send="true"><sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N></a><br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=17yv7m0rXBt8N<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.127.173></a><br>
Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784
New Channel sofia/exter<br>
<a moz-do-not-send="true"
href="mailto:nal/7806283672@66.51.127.163"
target="_blank">nal/7806283672@66.51.127.163</a>
[86c939d1-4de0-4a46-9203-518e0d6f7bc5]<br>
2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331
Processing CLID NAME<br>
<7806283672>->sipuser in context public<br>
2011-01-10 16:22:37.977477 [NOTICE]
switch_core_state_machine.c:189 sofia/extern<br>
<a moz-do-not-send="true"
href="mailto:al/7806283672@66.51.127.163"
target="_blank">al/7806283672@66.51.127.163</a> has
executed the last dialplan instruction, hanging<br>
up.<br>
2011-01-10 16:22:37.977477 [NOTICE]
switch_core_state_machine.c:191 Hangup sofia<br>
/external/<a moz-do-not-send="true"
href="mailto:7806283672@66.51.127.163" target="_blank">7806283672@66.51.127.163</a>
[CS_EXECUTE] [NORMAL_CLEARING]<br>
send 806 bytes to udp/[66.51.127.173]:5060 at
23:22:37.977477:<br>
------------------------------------------------------------------------<br>
SIP/2.0 480 Temporarily Unavailable<br>
Via: SIP/2.0/UDP
66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0<br>
Via: SIP/2.0/UDP
66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=17yv7m0rXBt8N<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.127.173></a>;tag=QU7286erpFDXm<br>
Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad<br>
CSeq: 7014814 INVITE<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f
2010-12-29 13-15-14 -06<br>
00<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
UPDATE, INFO, REGISTER, RE<br>
FER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, hold, refer<br>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>
Content-Length: 0<br>
Remote-Party-ID: "sipuser" <a moz-do-not-send="true"><sip:sipuser@192.168.35.1></a>;party=calling;privacy=o<br>
ff;screen=no<br>
<br>
------------------------------------------------------------------------<br>
2011-01-10 16:22:37.977477 [NOTICE]
switch_core_session.c:1273 Session 55 (sofia<br>
/external/<a moz-do-not-send="true"
href="mailto:7806283672@66.51.127.163" target="_blank">7806283672@66.51.127.163</a>)
Ended<br>
2011-01-10 16:22:37.977477 [NOTICE]
switch_core_session.c:1275 Close Channel sof<br>
<a moz-do-not-send="true"
href="mailto:ia/external/7806283672@66.51.127.163"
target="_blank">ia/external/7806283672@66.51.127.163</a>
[CS_DESTROY]<br>
recv 352 bytes from udp/[66.51.127.173]:5060 at
23:22:38.071224:<br>
------------------------------------------------------------------------<br>
ACK <a moz-do-not-send="true">sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1</a>
SIP/2.0<br>
Via: SIP/2.0/UDP
66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0<br>
From: "CLID NAME" <a moz-do-not-send="true"><sip:7806283672@66.51.127.163></a>;tag=17yv7m0rXBt8N<br>
Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad<br>
To: <a moz-do-not-send="true"><sip:14034883602@66.51.127.173></a>;tag=QU7286erpFDXm<br>
CSeq: 7014814 ACK<br>
Content-Length: 0<br>
<br>
------------------------------------------------------------------------
<div>
<div class="h5"><br>
<br>
<br>
On 10/01/2011 2:52 PM, Michael Collins wrote:
<blockquote type="cite">Or just give us your
credentials and we'll "test it thoroughly" for you.
:)
<div>-MC<br>
<br>
<div class="gmail_quote">On Mon, Jan 10, 2011 at
1:24 PM, Brian West <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:brian@freeswitch.org"
target="_blank">brian@freeswitch.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:
0pt 0pt 0pt 0.8ex; border-left: 1px solid
rgb(204, 204, 204); padding-left: 1ex;">can
you put up a sip trace or something so we can
help guide you?<br>
<font color="#888888"><br>
/b<br>
</font>
<div>
<div><br>
On Jan 10, 2011, at 2:26 PM, Darren Wiebe
wrote:<br>
<br>
> Good Afternoon,<br>
><br>
> I'm trying to get my freeswitch box
talking to Link2voip. Does anybody<br>
> have sample XML files for them?<br>
><br>
> --<br>
> Darren Wiebe<br>
> Aleph Communications<br>
> --------------------<br>
> Phone: 1-877-702-2900<br>
> Fax: 1-866-274-4506<br>
> Email: <a moz-do-not-send="true"
href="mailto:darren@aleph-com.net"
target="_blank">darren@aleph-com.net</a><br>
><br>
</div>
</div>
</blockquote>
</div>
</div>
</blockquote>
<br>
</div>
</div>
<font color="#888888">
<pre cols="72">--
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax: 1-866-274-4506
Email: <a moz-do-not-send="true" href="mailto:darren@aleph-com.net" target="_blank">darren@aleph-com.net</a></pre>
</font></div>
<br>
</blockquote>
</div>
</div>
</blockquote>
<br>
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