similar to: No DID on ZAP

Situation: 8 POTS lines, 3 companies, 1 system. Channel banking the POTS lines onto a T1 thru an ADIT 600. The only way our carrier will provide DNIS is thru Analog DID #'s. Anyone know of a piece of hardware that can add DNIS digits to a particular line? -Darren

Hello all, The "Secret Agent" final release of the Asterisk Management Portal is now available for download: http://amp.coalescentsystems.ca/ This exciting new release adds a great deal of functionality and flexibility. Thank you for all the contributions and feedback! 1.10.007 - Added AMP Users (multi-department, basic multi-tenant) - Added incremental upgrade script

Hello! I discovered, that I'm unable to load ther kernel module under 2.4.28. Before that I had 2.4.26 and tryed to upgrade the kernel to 2.4.28. After restart (reboot - soft restart) I can't load the module. When I go back to 2.4.26 and try to load the module, I'm getting the same problem. Only turning off/on helps. What can I do? It's very annoying because the box is in hosting

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

Guys. Maybe this is asking for a lot :) but is there any web client that can use IAX2 and ilbc? This is for a "call us" web idea.... Any leads?

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

As the title suggests, I was wondering if there was another source of documentation for Asterisk. Related: If one wanted to contribute to documentation, who would one contact? Thanks! Sean

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

Hi all! Somebody knows something to do with a high delay using Asterisk + DIAX!? When I used IAXComm(Linux) in both sides(peer and me) no problems. Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the voice coming from the person that I called. I don't have delay in my voice to the peer phone. CODEC: u-law (I tried with all available codecs) Thanks for your help!

Hi Can TDM400P detect polarity reversal on FXO module? We have C.O. lines that reverse polarity on Answer and release. Thank You This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

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

Good day all My extensions.conf is something like this [main] ;---incoming+ play welcome message extens => s...... ;---users extensions exten => 100..... ;---outgoing ignore 0 ;----------------- It all works fine The message says dial 1 for this ens.... But if I dial 0+number it will actually make a outgoing call! How do I stop this? I must allow the ignore 0 for internal uses but not if a

Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 --> FXS (TDM400P) --> Asterisk --> SIP GW --> PSTN --> Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with