<br><br><div class="gmail_quote">On Mon, Dec 29, 2008 at 1:55 PM, Brent Vrieze <span dir="ltr"><<a href="mailto:bvrieze@cimsoftware.com">bvrieze@cimsoftware.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I got no resonses to this and some funny bounces so I'm trying again.<br>
<br>
<br>
<br>
First of all Merry Christmas.<br>
<br>
Second, my first problem with my provider not staying registered with<br>
our server was my fault. We moved our server room and I restarted the<br>
test system and the production system causing them to ping-pong back and<br>
forth registering with our provider causing random problems, they are<br>
both set to register with the same account right now. I shut Asterisk<br>
down on the one and now we don't drop any longer. doh!!!<br>
<br>
Last, We are having DTMF problems with our provider (via:talk). Does<br>
anyone have any experience with them and if so can you share it?<br>
via:talk does have a sample sip.conf and extensions.conf file to use but<br>
the dial plan they set up does not require any DTMF so they may never<br>
have tested it. We have tried inband, auto, rfc2833 for our DTMF and<br>
nothing works. I have submitted a ticket with them but the last time I<br>
did that they never responded so that is why I am posting here.<br>
I signed up with another SIP provider for a test account and the DTMF<br>
passes no problem from them so I must conclude there is some setting<br>
that via:talk has that is causing the problem. via:talk will not<br>
confirm this but they must be using Asterisk as all the menus and such<br>
they have feel very Asteriskish. Is there something I can tell via:talk<br>
to try on their end to make this work?<br>
<br>
As a side symptem every time our system registers with via:talk it seams<br>
to jump from server to server on their end. They must have some sort of<br>
load balancing going on that is causing that. In the past we could get<br>
the DTMF to pass when we were on the initial server we registered with<br>
but when we got pushed to another server the DTMF would fail till I did<br>
a sip reload or restarted Astersk. Now we get no DTMF ever.<br>
<br>
System set up.<br>
Asterisk 1.4.22<br>
Asterisk GUI 2.0<br>
<br>
users.conf<br>
[trunk_1]<br>
context = DID_trunk_1<br>
host = <a href="http://galvatron.vtnoc.net" target="_blank">galvatron.vtnoc.net</a><br>
username = user name<br>
secret = password<br>
trunkname = via:talk - galvatron ; GUI metadata<br>
hasiax = no<br>
registeriax = no<br>
hassip = yes<br>
registersip = yes<br>
trunkstyle = voip<br>
hasexten = no<br>
fromuser = user name<br>
authuser = user name<br>
insecure = port,invite<br>
dtmf = rfc2833<br>
dtmfmode = rfc2833<br>
relaxdtmf = yes<br>
rfc2833compensate = yes<br>
port = 5060<br>
canreinvite = no<br>
fromdomain = <a href="http://galvatron.vtnoc.net" target="_blank">galvatron.vtnoc.net</a><br>
disallow = all<br>
allow = ulaw,gsm<br>
<br>
If you need to see more of the setup info I can provide.<br>
<br>
Thanks<br>
Brent<br>
</blockquote><div><br><br><br>I have the same problems with Viatalk. The problem is with their "new" servers. You are pointed to <a href="http://galvatron.vtnoc.net">galvatron.vtnoc.net</a> which is one of those. I currently have mine working by using their "old" servers. Try calling support, changing your account to rfc2833 if you haven't already and then point to <a href="http://chicago-1e.vtnoc.net">chicago-1e.vtnoc.net</a> with your same settings . You will have DTMF working, but I am not sure when the "old" servers are going away. <br>
<br>Good Luck,<br><br>Sean<br><br> </div></div><br>