Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ??
Where can I find it?
Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All,
Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
by Paul Durcek on Saturday 04 of March, 2006 [19:41:17 UTC]
I am not sure if it works in all situations, but I was able to get the actual dialed number information from the SIP header using the following command:
exten => s,1,NoOp(${SIP_HEADER(TO)})
Substrings
by jlewis on Monday 09 of January, 2006 [20:24:57 UTC]
Under Asterisk 1.0.x, substrings of the first N chars could be represented as ${variable::N}.
Under Asterisk 1.2.x, such substrings must be represented as ${variable:0:N}.
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Alternate for DNID function
exten => s,1,NoOp(${SIP_HEADER(TO)})
Substrings
Under Asterisk 1.2.x, such substrings must be represented as ${variable:0:N}.