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Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

...ence, INVITE -- Receives INVITE REDIRECT -- Sends 300 Successfully ACK -- Receives ACK But the actuall call is not redirected. Can anyone please help what is problem with the SIP redirection message and anyhelp to test this functionality please. You can download the source code from http://www.speak2world.com/asterisk/chan_sip.php Here is the procedure to compile and run it. 1. cd to /usr/src/asterisk/channels/ 2. Backup your existing chan_sip.c 3. replace the chan_sip.c with the current one 4. Type, "make install" when you receive a call, it should now pass the SIP 300 message to the c...

Is anyone selling G729 License elsewhere other than Digium? Anyone allowed to sell a similar License as a reseller? -Kannaiyan

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

I have this line in my extensions.conf, exten => _393.,1,Dial(ZAP/${EXTEN:3},20,tr) when I make a zap call, it gives me two rings and then makes the zap call. Is there is a way I can make the call immediate? Kannaiyan