similar to: How to use a data T-1?

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

I was just told that for my forthcoming system I will be getting a data T-1 instead of a voice T-1. Given that all of the handsets will be voip phones, no analog at all, do I need echo cancellation? I looked at the voip-info wiki and it seems to me that the answer should be "no" but I would like to confirm that. TIA, W

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the

Hi, Anyone using SpanDSP with Digium TDM o TE cards to receive and email Faxes? Thanks, Javier Ergas R. Director General de Tecnolog?a Pibix Telefon?a IP http://www.pibix.cl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060615/ed77e5d1/attachment.htm

How does one go about connecting Asterisk to a Meridian PBX to handle voicemail? I have a customer who is out of capacity on their voicemail system (which connects to their meridian via several FXS cards) and I would like to see if I could use Asterisk to handle their voicemail. -Jonathan

On Fri, Jul 20, 2001 at 11:27:16AM +0500, Asif M. Baloch wrote: > Hi guys, > > ICE cast resembles shoutcast, thats for sure and its good that its open > source. But, it has too many probs. I ran it on a dual p3 800 with a T3 comm We are running icecast on linux 2.2.19 serving more than 600gb each months and it has been rock stable so fare. Right now our uptime is around 3 months.

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to

We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode

I have been investigating more tight integration between * and the Nortel MICS... it appears that it is at least theoretically possible to have * store voicemail and log which stations call where. Both require a T1 card. The T1 card requires either a clocking module or the 6-port fiber module to provide T1 timing. Naturally a T100P or TE405P is required on the * side. To log which

Has anyone ever interfaced a merdian option 11 and asterisk. Just wondering how you went about, it's for a small setup me only need between 4/6 channels, I was thinking about using some spare ISDN channels between the two. Has anyone seen an SIP option for the meridian? ************************************************************ European Museum Of The Year 2002 The Chester Beatty

Hi all, I am going to be building an Asterisk system to replace the current aging (aged) Nortel Meridian system in a travel agency. There is already a voice T-1 in place and currently there are about 20 extensions in use. I would want to move up to about 25 extensions immediately and about 30-35 within the year. I am going to want IVR and voicemail, plus the ability to ring a group of

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

I have to connect a Meridian S1 to Asterisk for a slow migration to VoIP. What is the best way to connect them? 1. Is a T1 the best solution? 2. Can I pass Caller and Callee information across the link? If a T1 is best, I recall a standalone T1-SIP device a year ago on this list. Does anyone recall what the device was? How well it works? Thanks, MD -------------- next part --------------

at my work we have a meridian 1. it has 6 lines from the outside world coming in to it. if someone calls our number 870-238-2111 they will always reach our resepsionist unless all 6 lines are in use, if all 6 are in use then the caller will get a busy signal. on our phone bill they are called "trunk lines". we have a seperate incoming line for our dsl. are these analog lines ?? to

Okay, I'm terribly confused. If I build and run Asterisk with the 1.0.0 sources that I downloaded from Digium, my Polycom 300 works just fine. If I build with either various CVS builds, or the 1.0.6 sources from Digium, I get "No compatible codecs!". WTF? I'm using -the exact same- config files for both; I've tried enabling/disabling ULAW, ALAW and G279 on the

Has anyone integrated Asterisk for Voicemail with a Nortel Meridian phone system or know if this could be done? Thanks, John Haigh

Hi all, This is a NEWBIE question, so all you experienced types that are tired of stupid questions can move on... I've pretty much given up trying to do my entire phone system over IP (including local service), so I have to select and provision my local CO lines. I need about 10-12 lines which can be POTS lines, of course. But, I thought, why not get something digital and expandable