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My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

...sk-Users] Stability of Asterisk with 2 x > TDM400P cards (6analogue lines) (Krzysztof Drewicz) > 11. queue caveats (asterisk@anime.net) > 12. RE: Bluetooth headset in handsfree modewith SJPhoneor X-lite > (wendell hamilton) > 13. Re: Config File Management (Giovanni Miano) > 14. RE: Ability to put call on hold via manager? (Steve Totaro) > 15. Re: Authorization by ip (Giovanni Miano) > 16. Re: Call Simulator (Giovanni Miano) > 17. Re: Alarm on Unicall (Melcon Moraes) > 18. Re: Receptionist Phones (was 3Com Phones) (Justin Moore) > 19. R...

every few days my ADSL connection gets dropped for a few seconds. When it does I find my SIP connection to one of my providers does not timeout and retry. Does the following give some clues? Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others. (note this is the debian etch/testing package, I can build a new one if needed) .. CLI> sip show registry Host

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

Hi, we a have a SER (OpenSER) in front of 2 real-time Asterisk. SER simply forward SIP messages to 1 of the Asterisks: UA --> SER --> Asterisk We have a problem with REGISTERs: Asterisk answers with 200 OK, but changes the Contact header, inserting the IP of SER instead of the original IP (the IP of the UA). It seems that performs a sort of NAT-traversal, but all the elements are on

...= Spawn extension (default, 2093, 2) exited non zero etc etc I?m very stuck on this and can?t figure it out. Any help appreciated. Many thanks, Aisling. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Giovanni Miano Sent: 05 January 2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Incoming PSTN Calls Is Exist "InternalExtension" context ? and 2093 exten ? 2006/1/5, Aisling < ashling.odriscoll@cit.ie>: Hi all, I am having difficulty getting...

God save this mailling list :) Which is the best settings for the best quality of Audio ? I'use isdn4Linux driver and SIP client but bad quality PSTN <-> Asterisk and SIP <-> PSTN

I use codec g711u or g711a but comuncation between two sip client (XTen lite) have bastard dalay of 0,5 - 1 second Is it normal ? Are there any configuration to solve problem ? Thanks all

Hi, I've a billion isdn0 card but i suppose than i cant patch kernel to use mISDN support because i've kernel 2.4 and patch on CVS i4l is for >= 2.6.8 How do it ? help me please Thanks

Hi all, i have a question: what about a CTI implementation with Asterisk. I've been looking for info in www.voip-info.org <http://www.voip-info.org/> and in google, but There are no precise informations! Thanks a lot stefano -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Group, I am sending my question again why I don't have answer yet: I am developing a application, this use "Manager API" to connect with Asterisk. But when I call to an external number (over a zap channel), I don't receive any event when the target answer, Who can help me?, Which event notify me that the phone call was answered? Thank you. Ezequiel -------------- next

Hi, I'm testing Asterisk with MySql and I would want to insert sip users in a table "sip_users". After I modified extconfig.conf with "sipusers => odbc,asterisk" and I create the table sipusers, which changes must I make to sip.conf? Thank's Maury P.S.: C'? qualche utente italiano nella mailing list? -------------- next part -------------- An HTML attachment

Hi, I have built another Asterisk box using one ISDN HFC-S card and Bristuff-0.2.0-RC8p. But this time it behaves very strangely. Asterisk simply hangs and in logs I receive something like this: --NOTICE [1197]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 --NOTICE [1197]: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 And when I want to call a ZAP channel I get

...56f31y4986abc9054144 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. another thing what codec are you using? Date: Fri, 7 Dec 2007 17:02:31 +0000 From: "Giovanni Miano" <giomiano at gmail.com> Subject: Re: [asterisk-users] asterisk performance To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <d75be1ca0712070902u6d25ee49w368eda405a32bce8 at mail.gmail.com> Co...

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and