similar to: Using Asterisk as a TA?

I did post this a couple of days ago, but no-one has replied yet....so I thought I'd make the subject line a bit more accurate, add a bit, and try again! I've started looking at Asterisk as a possibility for a small PBX here. I'm thinking of an ISDN (BRI) card for connection to the telco, with some analogue converters (Sipura) for existing phones too. Once it's up and running

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

Hi, I have a terminal adaptor connected to Asterisk. I found that when it connect to Asterisk v1.0.0,The TA send register msg, and get OK msg, but it get Notify msg every 1 minute, The TA reply with "call does not exist" , usually 10 minutes later, The TA is down. But it work well with Asterisk v1.0-RC1, only register and OK msg are sent and got, no Notify msg is sent to it.

> -----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)

Hello. Has anyone been able to successfully make data (dialup modem) calls through a TDM2400E? We're able to make fax and credit card calls fine, but cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in

I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel:

Hello, with regard to the description of testing asterisk + USB ISDN TA on OpenSlug: http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk and NSLU2 running Debian http://peter.korsgaard.com/articles/debian-nslu2.php I've tested the same thing (asterisk as VoIP/PSTN gateway) on an NSLU2 running Debian (which uses the CPU in little endian mode). Results so far: - driver (mISND) compiles

Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a

I've been working with an adtran ta 850 hooked to a t100p pretty much all day today, and I haven't gotten past configuring zaptel.conf and zapata.conf. For some reason, when I pick up analog phone hooked up to the first module of a quad fxs card in the second slot of the ta 850, asterisk thinks that all four of the fxs modules in that card are going off hook. If I pick up a phone hooked

> I've got a Lucent TNT that I'm using for a gateway. Its working fine, but I > have one problem. I cannot find any place to set a ring timer, or number of > rings. The calls seem to timeout (Goes to all circuits busy) after about 15 > seconds - which isn't enough time for some voicemail boxes to pickup. I > found a setting called ringing-timer under sip-options, but

I sent this out about a week or so ago, but I have heard nothing. I am *thinking* it is a SELINUX problem, but I cannot figure out what. The SELINUX settings for both machines are *exactly* the same (the stock defaults for a standard CentOS 6 install). The *only* difference is that the desktop (sauron) has a few VMs setup (under KVM) and the laptop (gollum) does not. The desktop has an AMD

At Sat, 13 Feb 2016 10:14:30 -0500 CentOS mailing list <centos at centos.org> wrote: > > I sent this out about a week or so ago, but I have heard nothing. I am > *thinking* it is a SELINUX problem, but I cannot figure out what. The SELINUX > settings for both machines are *exactly* the same (the stock defaults for a > standard CentOS 6 install). The *only* difference is

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

I have not yet found a USB-to-serial adapter detected as /dev/ttyACM1. Try /dev/ttyUSB0 ? - Mike On Sat, Feb 13, 2016 at 10:14 AM, Robert Heller <heller at deepsoft.com> wrote: > I sent this out about a week or so ago, but I have heard nothing. I am > *thinking* it is a SELINUX problem, but I cannot figure out what. The > SELINUX > settings for both machines are *exactly* the

Has anyone managed to get Asterisk to work with any of the Alchemy or Cybergear ISDN switches? Cheers Matthew

The advice i was given was to spend the license money on a digium card and sell the dialogic on ebay! Whilst the digium card requires more from the host processor and may not be approved in as many countries as the equivelent dialogic, I think that for most cases the advice was sound. Actually, I kept the dialogic card, but that was for support purposes. Tim. Isamar Maia

Hi ALL, When i was accessing the voice message it suddenly goes dead and after that i couldn't able to retrieve the voicemessage again from my mailbox . This happens once in a while for any configured mailboxes I am using the following system configuration. asterisk 1.4.22.1 odbc storage of voicemail messages centos 5.2 64bit unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2

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