On 6/11/2008 2:06 p.m., Pedram M wrote: > Hi, > > Having some issues here with getting asterisk realtime for the dialplan > (extensions.conf) setup: > > mysql> desc extensions_table; > +----------+--------------+------+-----+---------+----------------+ > | Field | Type | Null | Key | Default | Extra | > +----------+--------------+------+-----+---------+----------------+ > | id | int(11) | NO | MUL | NULL | auto_increment | > | context | varchar(255) | NO | PRI | | | > | exten | varchar(255) | NO | PRI | | | > | priority | varchar(255) | NO | PRI | 0 | | > | app | varchar(255) | NO | | | | > | appdata | text | NO | | | | > +----------+--------------+------+-----+---------+----------------+ > > > ##################### > ### extconfig.conf file ### > ##################### > > extensions.conf => mysql,attributed,extensions_table > > > > Asterisk debug shows: > > > -- Attempting call on SIP/grnvoip/123804011818345XXXX for [EMAIL > PROTECTED]:1 > (Retry 1) > > == Starting SIP/grnvoip-09592260 at 10,start,1 failed so falling back to > exten 's' > > == Starting SIP/grnvoip-09592260 at 10,s,1 still failed so falling back to > context 'default' > > [Nov 5 19:04:42] WARNING[29109]: pbx.c:2470 __ast_pbx_run: Channel > 'SIP/grnvoip-09592260' sent into invalid extension 's' in context 'default', > but no invalid handler > > > This is with a call file that looks like: > > > Channel: SIP/grnvoip/123804011818345XXX > MaxRetries: 0 > RetryTime: 60 > WaitTime: 30 > Context: 10 > Extension: start > Priority: 1 > > > And in the database the context 10, extension start and priority 1 does > exist as shown below: > > mysql> select context,exten,priority,app from extensions_table limit 0,3; > +---------+-------+----------+----------------+ > | context | exten | priority | app | > +---------+-------+----------+----------------+ > | 10 | start | 1 | Set | > | 10 | start | 2 | AMD | > | 10 | start | 3 | WaitforSilence | > +---------+-------+----------+----------------+ > > > > Any ideas on where to begin w/ the debug would be very appreciated.
Are you doing the switch from the dialplan in the [10] context? Have a look at the realtime page on the wiki (voip-info.org) -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
