search for: snomsip

Hello list When i'm trying to dial into our pstn the following errors occure: -- Executing Dial("SIP/snomsip-dbd0", "/2100") in new stack Sep 9 10:02:22 WARNING[59409]: channel.c:1901 ast_request: No channel type registered for '' Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create channel of type '' == Everyone is busy/congested at this time --...

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

...icate on REGISTER to '<sip:phone@192.168.0.6>;tag=as3faa3b67' I have in sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to register => phone@192.168.0.6/phone ; 192.168.0.6 it?s my server linux ASTERISK. [snomsip] type=vozip secret=blah host=dynamic dtmfmode=inband defaultip=192.168.0.9 ; phone grandstream mailbox=1234,2345 restrictcid=no how can configure the extensions.conf if I want to sound my grandstream when I have incoming calls.? ANY IDEAS Cheers. vozi...

...; Allow codecs in order of preference ;allow=ilbc ;VoWLAN testing! allow=ulaw allow=alaw allow=gsm ;allow=all ; ;register => 1234@mysipprovider.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345 ; Mailbox for message waiting indicator ;restrictcid=yes ; To have the callerid restriced -> sent as ANI [30.0.0...