similar to: How can you keep agents logged in across a restart?

I hope someone can help me with this. This is what I want to happen. Someone dials in and goes to my extension. First, the phone on my desk rings If there is not an answer, I would like to have the dialplan call my cell phone. If I answer my cell phone, speak the incomming number to me. I press one of the buttons on my cell phone to accept the call. If I don't answer, or I don't

I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

Hi, I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is restarted. Can this be avoided in some way ? Regards, Rob.

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

Hey all again, I have successfully compiled and am running Asterisk (stable release) on OS X (10.3). However, any make directive that uses the "-shared" option in gcc results in an error. Apple states that -shared is not supported under OS X. Is there a workaround or do I have just have to live life without those modules (zaptel, libpri, format_mp3, probably others)? Thanks, Zach

> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens

Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible

Hello there, I am playing around with Asterisk the first time and it really looks great. ;-) However, I have one problem: Any SIP device can connect to my PBX. How can I requre logon for SIP users and deny access in the case of wrong or missing credentials? Thanks Florian

Hello All, I currently have an Asterisk Box, running a callcenter with 04 queues. I set queues.conf with "persistentmembers=yes" in the general section as follows: [general] monitor-type = MixMonitor persistentmembers = yes However when I perform any kind of restart in the Asterisk application, all agents are considered unavailable after that. Though when performing

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

Hi, I have several SIP registrations on my Asterisk box. Sometimes, I try to call in the inbound number from 1 and find it doesn't work. When I do sip show registry, it's showing Unregistered (and sometimes there are several which are showing Unregistered). If I type reload, it registers and works fine straight away. Is there something I can set to keep registrations connected all the