search for: january2001

Hi List.. I m getting this mesg while trying to dial an extension, both SIP UAs are registered with asterisk, m trying to dial extension 1015 from UA 12321@xyz.com to extension 1016 of UA 77777@xyz.com In extensions.conf I added exten => 1015,1, Dial(SIP/77777,20,tr) Any hint? JF WARNING[16397]: File pbx.c, Line 1153 (pbx_extension_helper): No application ' Dial' for

Hello Eric, call progress detection is the problem. Asterisk mistakenly recognizes the call to be answered and then still "hears" the ringing that should not be there if the line was really up. To solve the problem you would have to either implement a progress detection matching your country's indication tones or at least adjust the existing one for US or Costa Rica in dsp.c. By

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

We encouter a situation where we need to use SIP info to convey infomation for one end point to another endpoint. I use asterisk to do the test and find asterisk does not forward the SIP info to another endpoint, but act as UAS and returns a 4xx error message. I think asterisk is not right to handle this SIP info message. In RFC 3261 Page 70 "This protocol is designed to be extended.

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that