[SOLVED]SIP/Broadvoice outgoing setup
Posted: Wed May 28, 2008 12:01 amI 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
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.
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.