I installed astersik 1.2beta and from that point the led that indicate a new call flash. The ATA installed is an AZATEL. Any idea what can I check? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050921/040022d3/attachment.htm
Hi, I was not able to find any indication for this problem that I have right now. My phones connected to an Azatel 200 they always indicate that I have a message waiting to be listen. However, I do not have any message. I also checked using the console "show voicemail user for context" but I have 0 messages. Any idea what I need to check? I am using Asterisk 1.2.1 Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051225/553a5da8/attachment.htm
On Sun, 2005-12-25 at 12:28 -0500, C F wrote:> dial 911haha -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051225/e501c77a/attachment.pgp
Zeeshan
2005-Dec-25 10:51 UTC
[Asterisk-Users] s or _X. , is there any change since Asterisk 1.2
Hi, Before Asterisk 1.2 release, s extension never worked for my sip phone and I had to catch calls in my [incoming] using _X. but today after installing Asterisk 1.2, extension s is doing the same thing what _X. used to do. My understanding was that s extension was good only for FXO. Is there any change in its behavior since 1.2 that it is treating calls incoming on sip same as incoming on FXO. If so, which extension should be used and why? I am totally confused. Thanks Zeeshan A Zakaria
Zeeshan
2005-Dec-29 03:51 UTC
[Asterisk-Users] s or _X. , is there any change since Asterisk 1.2
Hi, Before Asterisk 1.2 release, s extension never worked for my sip phone and I had to catch calls in my [incoming] using _X. but today after installing Asterisk 1.2, extension s is doing the same thing what _X. used to do. My understanding was that s extension was good only for FXO. Is there any change in its behavior since 1.2 that it is treating calls incoming on sip same as incoming on FXO. If so, which extension should be used and why? I am totally confused. Thanks Zeeshan A Zakaria