similar to: Error Message

Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06

Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? ________________________________________________________________ Anton Krall Direccion General Intruder Consulting A Division of IntruderEnterprises S.A. de C.V. www.Intruder.com.mx www.IntruderStore.com.mx Tel. 3872-2200 ext. 201

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

I gave a queue setup like this, but I also have it setup so that if no agents are online, the caller cannot get in but I discovered that if that's the case, the call hangsup on the caller: [soportetecnico] ;Soporte Tecnico exten => 2,1,Playback(${SONIDOS}/transferringcall) exten => 2,2,Queue(Soporte-Tecnico) exten => 2-.,1,Playback(noagents) I want to play a message tothecaller

After I leave a voicemail for an extension and hangup, my asterisk console (with debug turned up quite high) shows two error messages like: WARNING[7664]: app_queue.c:341 changethread: Can't change device with no technology! WARNING[7668]: app_queue.c:341 changethread: Can't change device with no technology! Clues? Thanks! -Dorn

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

Hello, We use astGUIclient, it does have server side apps that have to be installed on your Asterisk server, but it does have callerID popups that allow you to search a customizable web page when a call comes in. We are also releasing a new version of the astGUIclient app next week that is entirely web-based and easier to configure the client side. http://astguiclient.sf.net/ MATT---

Guys. What does this error mean? -- Playing '/var/spool/asterisk/voicemail/intruder/201/unavail' (language 'sp') -- Playing 'vm-intro' (language 'sp') -- Playing 'beep' (language 'sp') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/intruder/201/INBOX/msg0000 format: wav, 0x812b4f0 -- User ended

Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __________________________________________________________________ Anton Krall

I have a weird problem... very puzzling.. Yesterday I had sound problems with the voice prompts, I couldnt hear them, so I rebooted the system and voila, I was able to hear everything.. so I went to bad.. and I just woke up and tried the system again and its back!!! I dial the voicemail system and I cant hear the voice welcome.. I can hear any voice prompts.... Has anybody had this kind of

Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? __________________________________________________________________ Anton Krall

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

I'm running Asterisk HEAD from March 4. I've Googled a bit but I can't figure out what causes this error: app_queue.c:374 in changethread: Can't change device '**Unknown**' with no technology! It doesn't seem to be causing any problems, but I'm curious what causes it. I did a few Google searches and found a lot of people asking about it, but no real answers.

hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 > Send asterisk-users mailing list submissions to > asterisk-users at

Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS devide

The SC420 does not support APIC (at least not with current BIOS). The Digium TDM400B card seems to always seek the same IRQ so it locks one down. The clone X100P cards often get a shared IRQ, even if vacant ones are available. I've found both the Digium card and the clones very frustrating with this system. Thus the reason Digium does not recommend it. I've gone to a mostly SIP

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences..

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