[Asterisk-Users] SIP Phone dropping calls, SIP Softphones working fine

[etx]

Hello! 

I am semi-new to asterisk, I've been toying with it for about a month
now. I'm using the current version from the CVS server (as of last
week) and have succesfully connected it to Broadvoice, as well as a
few softphones. I got a Nortel i2004 IP phone from a friend to setup
at home as a homephone using Broadvoice, but going through my * box
and using iptel.com for toll free calls. Well everything is basically
working with the exeption of one thing, outgoing calls from the Nortel
i2004 drop as soon as the remote party answers. For example if I call
my celular phone it will ring, but as soon as I answer it the call
will drop. It first I thought it was * tring to hand the call off so
that it would go directly from Broadvoice to the SIP phone. I'm still
not sure, I've tried different codecs in the sip.conf, and I've tried
canreinvite=no in the peer definition. One strange thing I found is
while seting up music on hold I created an extention to test it. I
forgot to add an Answer command for that extention, so it would just
start playing the MOH music and I could hear it on the nortel i2004.
After adding the Answer command it started droping the call. Also,
this could all be due to the simple fact that this is a nortel phone.
It does not natively support SIP, I have to use the Nortel Multimedia
PC Client. It will connect to the phone and bridge everything to the
sip server I configure on the PC. It just uses the username/pwd, sip
proxy server, sip proxy port, and the domain. I have the Firefly
softphone working flawlessly, but a softphone is not going to cut it
at home. Here is some verbose output from * when I try to make the
outgoing call. Again, Incoming calls to the i2004 work perfectly. I'm
stumped.

Connected to Asterisk CVS-HEAD-09/12/04-17:37:58 currently running on
Barrat (pid = 1086)
    -- Executing Goto("SIP/i2004-a783", "outgoing|13136717890|1") in new stack
    -- Goto (outgoing,13136717890,1)
    -- Executing Goto("SIP/i2004-a783", "pri-out|13136717890|1") in new stack
    -- Goto (pri-out,13136717890,1)
    -- Executing Dial("SIP/i2004-a783", "SIP/13136717890 at broadvoice")
in new stack
    -- Called 13136717890 at broadvoice
    -- SIP/broadvoice-9afd is making progress passing it to SIP/i2004-a783
    -- SIP/broadvoice-9afd answered SIP/i2004-a783
Sep 21 11:31:49 NOTICE[98310]: chan_sip.c:7519 handle_request: Unknown
SIP command 'PING' from '69.14.242.210'
Sep 21 11:31:50 WARNING[98310]: chan_sip.c:680 retrans_pkt: Maximum
retries exceeded on call cf27d_ff22b1386c at 192.168.0.106 for seqno 1
(Non-critical Response)
Sep 21 11:31:59 NOTICE[98310]: chan_sip.c:7519 handle_request: Unknown
SIP command 'PING' from '69.14.242.210'


Thanks for any help you could provide! And Thanks for developing such
a nice open source PBX!