similar to: Lines Not being Hung UP Major

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 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 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 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

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

I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been

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

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

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 all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. That being said, they are also alot more money than the Rhino cards and my friend currently has 1 digium 4 fxo card in their system and they need to add another phone line, plus they have echo problems and quality

---------- 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, 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

Hi, Sorry for keep hammering the list with this annoying question. Can we use Intel 536 ep (not 537ep that is in wiki) as x100p clone? I know I've asked it in this list a couple days ago but none responded so far and I'm getting frustrated pairing it with asterisk as the zaptel driver could not detect it. I just need more information before I throw this intel 536 EP to the garbage can

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

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

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actual measured at 8.86 mbps. The

Hi, I want to pick up a call with the snom's programmable buttons(snom190 -SIP 3.60x, snom360-SIP 4.1) with asterisk server (v 1.2.0), I tried with the option 'Destination' and when the incoming call arrive to another snom phone the button blinking. In this way I can only pick down it pressing the blinking button. The solution is call the *8 or parcking the call but my pbroblem

Hello, I have read the articles on echo cancellation (http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a solution to my problem. We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M PCI (current driver from EICON) and some SNOM 300/360. There are few clients where we recognize echoes on both sides when we call them via ISDN. With some of these clients we