similar to: Thinking of moving from pure VoIP to PRI - thoughts?

Hi Guys, Running the following with a Sangoma A101D PRI card: *Asterisk 1.4.21.2* *LibPRI version: 1.4.10* No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show any activity. Problem goes away on restart of the system or maybe asterisk. I see post about upgrading Libpri to 1.4.10.2 and then I see posts that even that didn't work. Anyone can weigh in this please?

Hi list, My Asterisk config for outgoing calls is the following: exten => s,1,Dial(SIP/${MACRO_EXTEN}@voipprovider,60,g) exten => s,n,GotoIf($[\"${ANSWEREDTIME}\" = \"\"]?pri:hang) exten => s,n(pri),NoOp(Problems with voip provider trying PRI) exten => s,n,Dial(Zap/g2/${MACRO_EXTEN},60,g) exten => s,n(hang),HangUp in most cases it works well but, if my

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Hi everyone, I'm new to the list, so I apologize if this is a trite question. We're an ISP with a barely used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. We have a notion that we want to take on a new revenue stream, with our model being as such: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece

I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that "true" ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also

Hi, Does anyone know if it's possible to have 1 port of TE406P to provide T1 PRI to other VoIP gateways? I am trying to provide T1 PRI to one of our old Clarent gateways to integrate it to Asterisk. Can anyone help? Thank you. Andy

Hi, I'm looking for HW to terminate VOIP calls (SIP, if IAX2 even better) from remote Asterisks on PRI EURO ISDN interface. One option is PC with Asterisk and PRI card, but are there any embedded devices to do that ? Will they be at same price or more expensive ? Regards, Rob.

We have been running around than 40 asterisk servers running on Debian Squeeze for last three years, handling traffic of more than few hundred thousand calls per day. Our setup's PRI-banks were using Redfone's Fonebridge. We had PRIs from multiple telephony providers. And Redfone's Fonebridge handled all that easily. But all good things come to end. Redfone's Fonebridge was not

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with

I'm putting together a PBX using a TE420P card configured for E1s that is connected to an Errickson MTS. successfully compiled and installed libpri 1.4.11.3, DAHDI 2.3.0.1+2.3.0 and Asterisk 1.6.2.9. everythings seems to be working. SIP phone to SIP phone (POLYCOM) calls work fine however, network calls do not. When I went to debug PRI, the only command that showed up when I did ">CLI

I have a friend with a PRI coming into a modem bank that is receiving 56K modem calls and some ISDN data calls. He wants to dump his analog office phone lines and use some of the capacity on the PRI. I have been digging through the mail archives and Wiki site on this subject but the information I found doesn't give me a high enough confidence to go buy a few T1 cards and try it out.

Hi, I'm buying an interface for my Toshiba DKT424 to connect to *. Initially it is a trial to route internal calls over our wan. Think of it as a tie trunk between 2 pbxs. I expect to get PRI from the telco our layout will be like this: telco_T1_PRI to * to various PBX or TDM or IAX to * to other pbxs My question is, is there benefits to connecting with PRI over a standard T1 within my

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

We are in the process of ordering a Voice PRI to plug into Asterisk. Of course we will be buying a card from Digium for this. Question is this, there seem to be MANY options technically when ordering this PRI (in the US) but since this is the first time ordering a voice circuit I am clueless as to what options we need. Any clues would be helpful or maybe something has already been written

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hi List, I have configured TE121PF card in E1 mode. I am using asterisk 1.6 and freepbx 2.7. My card staus turn in to green and looks like sync with the service provider. My service provider is BSNL - India. I have one toll free number for incoming and one land line number for out going calls. Problem : If i am calling to the toll free number, i am getting the ring but that call is