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.

2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME
 <7806283672>->sipuser in context public

What am I missing?

Here's the relevant provider entry from the external sip profile

<include>
  <gateway name="link2voip1">
  <param name="username" value="sipuser"/>
  <param name="password" value="password"/>
  <param name="proxy" value="sip.ca1.link2voip.com"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="udp"/>
  <param name="retry-seconds" value="30"/>
  </gateway>
</include>

Sip Trace: 

   From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as01e3f5de
   Call-ID: 0472a7de73e00fc578b12f6679b6dd9d@CUSTOMERIP
   To: <sip:14034883602@sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H
   CSeq: 103 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027:
   ------------------------------------------------------------------------
   INVITE sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
0
   Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
   Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
   From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd
   To: <sip:14034883602@sip.ca2.link2voip.com>
   Contact: <sip:7806283672@CUSTOMERIP>
   Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP
   CSeq: 103 INVITE
   User-Agent: Asterisk PBX
   Max-Forwards: 69
   Date: Mon, 10 Jan 2011 23:22:36 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Content-Type: application/sdp
   Content-Length: 235

   v=0
   o=root 12790 12791 IN IP4 CUSTOMERIP
   s=session
   c=IN IP4 66.51.110.210
   t=0 0
   m=audio 14648 RTP/AVP 0 96
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 telephone-event/8000
   a=fmtp:96 0-16
   a=silenceSupp:off - - - -
   a=nortpproxy:yes
   ------------------------------------------------------------------------
send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
   Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
   Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd>
   From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd
   To: <sip:14034883602@sip.ca2.link2voip.com>
   Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP
   CSeq: 103 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Content-Length: 0

   ------------------------------------------------------------------------
2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New Channel sofia/exter
nal/7806283672@CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab]
2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME
 <7806283672>->sipuser in context public
2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:189 sofia/extern
al/7806283672@CUSTOMERIP has executed the last dialplan instruction, hangin
g up.
2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:191 Hangup sofia
/external/7806283672@CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING]
send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
   Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060
   From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd
   To: <sip:14034883602@sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
   Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP
   CSeq: 103 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE
FER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1>;party=calling;privacy=o
ff;screen=no

   ------------------------------------------------------------------------
2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273 Session 53 (sofia
/external/7806283672@CUSTOMERIP) Ended
2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275 Close Channel sof
ia/external/7806283672@CUSTOMERIP [CS_DESTROY]
recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
   ------------------------------------------------------------------------
   ACK sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.0
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0
   From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd
   Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP
   To: <sip:14034883602@sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
   CSeq: 103 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983:
   ------------------------------------------------------------------------
   INVITE sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.
0
   Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
   Max-Forwards: 66
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e
   To: <sip:14034883602@66.51.110.210>
   Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   Contact: <sip:ciscosip@66.51.127.163:5080>
   User-Agent: Cisco-SIPGateway/IOS-12.x
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 323
   Remote-Party-ID: "CLID NAME" <sip:7806283672@66.51.127.163>;screen=yes;pri
vacy=off

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072 338127983379325658 IN IP4
66.51.127.163
   s=SIP Call
   c=IN IP4 66.51.110.210
   t=0 0
   m=audio 14650 RTP/AVP 0 18 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   a=nortpproxy:yes
   ------------------------------------------------------------------------
send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
   Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e>
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e
   To: <sip:14034883602@66.51.110.210>
   Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Content-Length: 0

   ------------------------------------------------------------------------
2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New Channel sofia/exter
nal/7806283672@66.51.127.163 [86b78ecd-469f-4a1c-9fe5-692a5941ff37]
2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME
 <7806283672>->sipuser in context public
2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:189 sofia/extern
al/7806283672@66.51.127.163 has executed the last dialplan instruction, hanging
up.
2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:191 Hangup sofia
/external/7806283672@66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK0D66gH7m2614F
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e
   To: <sip:14034883602@66.51.110.210>;tag=pjea7BymS6paS
   Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE
FER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1>;party=calling;privacy=o
ff;screen=no

   ------------------------------------------------------------------------
2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273 Session 54 (sofia
/external/7806283672@66.51.127.163) Ended
2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275 Close Channel sof
ia/external/7806283672@66.51.127.163 [CS_DESTROY]
recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979:
   ------------------------------------------------------------------------
   ACK sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2 SIP/2.0
   Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e
   Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
   To: <sip:14034883602@66.51.110.210>;tag=pjea7BymS6paS
   CSeq: 7014814 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477:
   ------------------------------------------------------------------------
   INVITE sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1 SIP/2.
0
   Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
   Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
   Max-Forwards: 66
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N
   To: <sip:14034883602@66.51.127.173>
   Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   Contact: <sip:ciscosip@66.51.127.163:5080>
   User-Agent: Cisco-SIPGateway/IOS-12.x
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 323
   Remote-Party-ID: "CLID NAME" <sip:7806283672@66.51.127.163>;screen=yes;pri
vacy=off

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778 787843793424096957 IN IP4
66.51.127.163
   s=SIP Call
   c=IN IP4 66.51.127.173
   t=0 0
   m=audio 15488 RTP/AVP 0 18 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   a=nortpproxy:yes
   ------------------------------------------------------------------------
send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
   Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N>
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N
   To: <sip:14034883602@66.51.127.173>
   Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Content-Length: 0

   ------------------------------------------------------------------------
2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New Channel sofia/exter
nal/7806283672@66.51.127.163 [86c939d1-4de0-4a46-9203-518e0d6f7bc5]
2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME
 <7806283672>->sipuser in context public
2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:189 sofia/extern
al/7806283672@66.51.127.163 has executed the last dialplan instruction, hanging
up.
2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:191 Hangup sofia
/external/7806283672@66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING]
send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477:
   ------------------------------------------------------------------------
   SIP/2.0 480 Temporarily Unavailable
   Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
   Via: SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N
   To: <sip:14034883602@66.51.127.173>;tag=QU7286erpFDXm
   Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
   CSeq: 7014814 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06
00
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE
FER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1>;party=calling;privacy=o
ff;screen=no

   ------------------------------------------------------------------------
2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273 Session 55 (sofia
/external/7806283672@66.51.127.163) Ended
2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275 Close Channel sof
ia/external/7806283672@66.51.127.163 [CS_DESTROY]
recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224:
   ------------------------------------------------------------------------
   ACK sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1 SIP/2.0
   Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0
   From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N
   Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
   To: <sip:14034883602@66.51.127.173>;tag=QU7286erpFDXm
   CSeq: 7014814 ACK
   Content-Length: 0

   ------------------------------------------------------------------------


On 10/01/2011 2:52 PM, Michael Collins wrote:
Or just give us your credentials and we'll "test it thoroughly" for you. :)
-MC

On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian@freeswitch.org> wrote:
can you put up a sip trace or something so we can help guide you?

/b

On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:

> Good Afternoon,
>
> I'm trying to get my freeswitch box talking to Link2voip.  Does anybody
> have sample XML files for them?
>
> --
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax:   1-866-274-4506
> Email: darren@aleph-com.net
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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-- 
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax:   1-866-274-4506
Email: darren@aleph-com.net