similar to: Queues Not Reporting Estimated Hold Time

hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the

Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue

Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and

A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. Basically i am looking to get some new phones and in the process get rid of the countless number of

Hi, I am getting a new sangoma t1 card soon and that will max out my slots, which means i need to take out a card. I am going to take out my pci network interface card (10/100) I have an open pci-e slot i have never used in the machine so i am going to buy a pci-e 10/100 or gigabit network adapter. I want to find one that works natively with the linux kernel. I hate using hardware that

Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful

Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Is anyone using the SNOM 360 as a reception console with Asterisk? We are trying to have the ability to view whether an extension is on or off hook, or ringing with the Snom, which seems to work fine. The issue is that we are having difficulty picking up calls and transferring. Anyone have experience / insight? Darrell S. Long Director of Technology BestWeb Corporation Phone 877-777-2932

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi Folks, Can anyone give me some good recommendations for VoIP providrs that support Asterisk PBX's? We're based in Georgia and I having a hard time finding anyone.... Regards, Jim PS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.com

Hi, i am having a problem with my point to point t1, which is being resolved and is a seperate issue. sangoma support has been a huge help and i am waiting on verizon to increase the signal output of the smartjack. But my issue is that in the meantime my fallover extensions arent working. Well they are on the CPE side but not on the NET side. The NET side still thinks its making calls, they

Hey, I got a weird one for you guys, I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

As recommened I got the new firmware for my echo cancellers and it solved hte problem with the agressive echo cancelling causing half duplex audio. I have to say, so far these cards are far superior to the previous models. The sound quality is hugely improved (enough to really notice which is alot) and the echo canceller works way better than the software ones. My system seems to like these

Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset's speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't affecting

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

FYI, spoke with Grandstream this morning, the GXP-2000 release has been delayed again. Looking like April now before these hit the street. -- Cory Andrews Senior Partner VOIPSupply.com +++++++++++++ V: 800.398.VOIP X22 F: 716.630.1548 E: Cory@VOIPSupply.com

Hi all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with the point 2 point t1. I am currently at the location trying to set things up but have some