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[SOLVED]SIP/Broadvoice outgoing setup

PostPosted: Wed May 28, 2008 12:01 am
by patg
I am trying to setup a test server with a single broadvoice trunk. Inbound and outbound calling work directly from/to extensions. When attempting to dial out from the astguiclient I get the following console output.


I would post the complete cli output but I cannot get it past the forum "new account" filter even with excessive editing.


May 28 00:23:23 WARNING[16808]: channel.c:2621 ast_request: No channel type registered for 'zap'
May 28 00:23:23 WARNING[16808]: app_meetme.c:480 build_conf: Unable to open pseudo channel - trying device
May 28 00:23:23 WARNING[16808]: app_meetme.c:483 build_conf: Unable to open pseudo device
-- Playing 'conf-invalid' (language 'en')
== Starting Local/8600051<at>default-f109,1 at default,<number>,1 failed so falling back to exten 's'


I have setup the following

Code: Select all
extensions.conf
SIPtrunk=<phonenumber><at>sip<dot>broadvoice<dot>com:PASSWORD:<phonenumber><at>sip<dot>broadvoice<dot>com
also tried
SIPtrunk=SIP/<phonenumber><at>sip<dot>broadvoice<dot>com:PASSWORD:<phonenumber><at>sip<dot>broadvoice<dot>com


This is the same string that I use in sip.conf to register the trunk.

I have also set this value in the "Outbound Call Group" and set "Client Protocol" to SIP under the phone configuration.

When attempting to do a manual dial from the astguiclient the asterisk cli output shows "No channel type registered for 'zap' " I do have a PRI card in this box as it will eventually be connected via PRI. I have tried unloading all zaptel modules and issuing an "unload chan_zap.so". Despite these settings I continue to get the zap warnings, do they matter?

I am pretty much stuck. I would really like to get this system tested prior to the PRI turn-up. Any suggestions on how to properly configure this system with a SIP trunk?

One other issue is I do not want the 1 dialed before the number, perhaps this is the whole issue, as there are no dial plan rules to handle a 1 before the number and it is not required. I have set the following under the campaign details.

Dial Prefix:X
Omit Phone Code: Y

The documentation on this seems clear enough but the result does not seem to be what I expect.

PostPosted: Thu May 29, 2008 11:24 pm
by patg
For broadvoice this is actually fairly simple.

Just follow the broadvoice setup instructions in your sip.conf

In extensions.conf set

Code: Select all
SIPtrunk=sip<dot>broadvoice<dot>com


and in the dialing context(s) use

Code: Select all
dial(SIP/${EXTEN}@${SIPTrunk}...


The example extension.conf with

SIPtrunk=SIP/1234:PASSWORD@sip<dot>provider<dot>net

is not the correct format for broadvoice