search for: astuser

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2006 May 03
4
QSIG support in Asterisk
I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060503/3cb7f966/attachment.htm
2007 Jul 27
1
Asterisk Users Conference Friday at 12:30 PM EDT
...services or products your company provides and answer users' questions, contact me off list. Anyone is welcome to be a guest and answer users' questions. Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox, Adhearsion Listen to the archived recordings here: http://x2z.eu/astusers.htm The Asterisk Users Conference is independently run and has nothing to do contractually or financially with Digium who owns the Asterisk trademark. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070...
2008 Jan 04
3
Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC
TODAY, Friday January 4th at 12 Noon EST, 11 AM Central, 9AM Pacific, Mountain figure it out, 17:00 UTC Mark joins us to talk about IAX, the appliance, what's new in the asterisk worldwide communities and answer any questions you may have. Why not take this opportunity to ask questions or make comments? This conference is the largest *live* online meeting of asterisk users in the world. Each
2008 Jun 27
1
Asterisk, POTS and plain handsets
Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook,
2007 Jan 28
1
Voicemail from sip phones
Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just
2007 Jun 09
0
Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone
Oops, I had some problems and was offline unable to remind you about the conference yesterday. LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash player, will autostart) THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the new book, whick looks like a great effort. A surprise visit from Jeremy, one of the pioneers of our community who started Nufone when someone on IRC said "I need a new phone". SI...
2007 Jul 27
0
Asterisk Users Conference Friday at 12:30 PM, EDT
...gt; and answer users' questions, contact me off list. Anyone is welcome to be a > guest and answer users' questions. > > Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox, > Adhearsion > > Listen to the archived recordings here: > > http://x2z.eu/astusers.htm I would certainly recommend that people participate in this conference. It is more than "a call" or podcast, it is a forum where we can come together to discuss issues, ask questions, and be part of the community. If you want to discuss any topic related to Asterisk or if you sim...
2007 Sep 06
1
Asterisk Users Conference Friday @ 12:30PM EDT
FRIDAY September 7th at 12:30 PM EDT http://www.asteriskusersconference.org for more information on how to listen, talk, or both :) This week, ENUM is the main subject, although our friends at e164.org haven't been able to talk to us as planned. Come on by and share what you know about ENUM or ask questions. Also, during Astricon, we are hoping people will call us with reports, either live
2007 Nov 06
1
Sangoma S200 and Digium TDM400P together
Hi, I have these two cards, the Sangoma has 4 fxo interfaces and the digium has 1 fxo and 1 fxs. After install the sangoma card, my zaptel.conf was configured for that card. I'm trying to configure the Digium one together thinking that the Digium ports should be 5 and 8 but it doesn't works. Someone has some example about this? Thanks in advance Pau?p
2008 Apr 18
0
Friday @12Noon EDT: VoIP Users Conference on the Internet
....212.134.192, 60, D(22622# ${MY_PIN} #) ) If you have no PIN use 1# instead. (remove any spaces in the line above) INFO: http://www.VoipUsersConference.org http://food4wine.ning.com IRC: Follow chatter or ask questions on IRC on Freenode.net #voip-users-conference RSS: http://feeds.feedburner.com/AstUser /r
2009 Aug 11
0
FSK UK Problems
...g FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about this. Thanks in advance. -- -------------------------------------------- Paulo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090811/cae3732c/attachment.htm
2007 Jul 19
4
Why using usecallerid=no?
Hi everybody. I'm in a discussion and someone ask me in which situation we should use the zapata.conf usecallerid set to no. I didn't have the answer. I understand what the usecallerid keyword does but I'm talking about a actual situation that is interesting to to avoid receiving the caller id. Thanks in advance! -- -------------------------------------------- Paulo Garcia Pika
2009 Feb 07
0
Asterisk 3rd party developed commercial software sales licensing platform
...rence/browse_thread/thread/6fe7d4af7ecdd996%3Fhl %3Den> @66.212.134.192, 60, D(22622# ${MY_PIN} #) ) If you have no PIN use 1# instead. (remove any spaces in the line above) IRC: Follow chatter or ask questions on IRC on Freenode.net #voip-users-conference RSS: http://feeds.feedburner.com/AstUser INFO: http://www.VoipUsersConference.org http://food4wine.ning.com For those of you who have never participated before make this your chance to get involved, download the talkshoe chat application in advance or even better go and listen to some of the previous 80 calls archived in mp3...
2007 Jul 12
0
No subject
...ce/browse_thread/thread/6fe7d4af7ecdd996%3Fh= l %3Den> @66.212.134.192, 60, D(22622# ${MY_PIN} #) )=20 If you have no PIN use 1# instead. (remove any spaces in the line above) IRC: Follow chatter or ask questions on IRC on Freenode.net #voip-users-conference=20 RSS: http://feeds.feedburner.com/AstUser=20 INFO:=20 http://www.VoipUsersConference.org=20 http://food4wine.ning.com=20 =20 For those of you who have never participated before make this your chance to get involved, download the talkshoe chat application in advance or even better go and listen to some of the previous 80 calls archived i...
2010 Sep 17
2
3rd party app store
...nference/browse_thread/thread/6fe7d4af7ecdd996%3Fhl %3Den> @66.212.134.192, 60, D(22622# ${MY_PIN} #) ) If you have no PIN use 1# instead. (remove any spaces in the line above) IRC: Follow chatter or ask questions on IRC on Freenode.net #voip-users-conference RSS: http://feeds.feedburner.com/AstUser INFO: http://www.VoipUsersConference.org http://food4wine.ning.com For those of you who have never participated before make this your chance to get involved, download the talkshoe chat application in advance or even better go and listen to some of the previous 80 calls archived in mp3 format...
2007 Sep 17
7
Why does everyone seem to dislike *now?
Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few
2006 Jun 06
0
pbx_spool - outgoing qcall failure upon call progress
Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and
2006 Jun 07
1
TBCT - Two B-Channel Transfer
Hi- A customer of mine running asterisk has inquired if asterisk currently supports TBCT, that is the ability of asterisk to transfer a call back up to the carrier's switch to complete a connection, freeing up the B-channels on the asterisk box. I saw a reference to this feature in the asterisk "bounties" section on the Wiki, and someone added a note saying that this might be
2007 Jun 15
0
FXS card with 3-way call, transfer and call waiting.
Hi, I would like to understand how those features (subject) work on fxs ports. Unfortunately I don't have a digium card with this kind of port, then any help will be appreciated. I tried to gather some information from google and this list history, but I still need some help. 3-way-call - As I could understand, when you are talking to A-person, you can press *flash*, call to B-person and
2007 Jul 01
0
Rockwell ACD - "Take back and transfer"
Hi all- I have a customer with a Rockwell Spectrum ACD. They wish to connect to an asterisk system using euro-ISDN circuits, and asked me if asterisk supports a feature that they call "Take back and Transfer". This feature allows an IVR (asterisk in this case) to handle a call and then blind-transfer it back through the ACD to another extension- freeing up the asterisk channel.