similar to: DTMF pass-through question

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- -------------- next part -------------- An HTML attachment was scrubbed... URL:

I currently have Asterisk deployed in my office with a TE411P. On the first port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm not receiving caller ID on inbound calls from this line. The caller ID information is arriving in the form *ANI*DNIS*. In zapata.conf, I have signalling set to em_w. The DNIS always arrives correctly, but I'm never receiving the ANI

Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried

Hello, We've released another update to our astGUIclient suite: 2.0.2 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk

Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an

Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

Hello, Now that we've had our new Digium TE406P card in production for 4 days we have discovered audio drop problems that happen randomly across all channels. Here's more about our setup: P4-3.2GHz 2GB ram Slackware Linux 10.1 with custom kernel 2.4.29 Asterisk 1.2beta1 Digium TE406P quad T1 card with the following attached: - 2 x RBS D4/AMI 24 channel T1s - 1 x RBS B8ZS/ESF 24 channel

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Is there a way to dial DTMF after bridging the call. The current option D() in Dial will dial DTMF before the call is bridged and this doesn't do the job. I need to dial DTMF after the call is bridged and the message is played with "Background" -- #Joseph

So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and

Hi All I cannot seem to find a way to stop atserisk inercepting DTMF tones and regenerating them even on a zap to zap bridged call is this possible? Thanks Robb

Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap