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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

...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/200707...

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

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,

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

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". SIP...

...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 simp...

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

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

Hi, I'm not sure at this point who will be with us, but there's always something to talk about on the conference, Fridays at 12 Noon Eastern Time, 9 AM Pacific, 4PM GMT. I have invited Garrett Smith whose blog you should consider following. PSTN: (724) 444-7444 Call ID: 22622 SIP: exten => 1234, 1,Dial(SIP/123 at 66.212.134.192, 60, D(22622# ${MY_PIN} #) ) If you have no PIN use 1#

Hi, I'm currently having problems detecting 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

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

I think it's time to 'ping' John Todd and Digium on this topic again. What happened Digium? Why did you say you were going to take this project on but have not come back to the community with an answer yet? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial).

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

I recently came across this email that I wrote in May 2008 ...... ?http://lists.digium.com/pipermail/asterisk-users/2008-May/210887.html It's such a shame that Digium manhandled the project away from the community only to then bury it and not allow it to proceed. I really wonder when I look at the Apple iphone development community as to where the 3rd party Asterisk development community

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

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

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

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

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.