similar to: Asterisk @ Home Voicemail

All, I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones. All seems well other than the phones have to be reset up to 5 times per day. It is like they lose thier ip connection or maybe thier SIP connection. Has anyone else experienced this issue? I have the phones set for static IP addresses and that doesnt seem to help either. Any help would be greatly

OK, now this should be really simple, but I am a bit of a newbie so please bear with me. I have an *@Home box setup with TDM04B and two POTS lines. On the SIP side, I have GXP2000 phones. Most things seem to work, but the users cannot figure out how to transfer incoming calls from one extension to another. Now I am not sure that I have things setup correctly, but is there something

I would like to install a TDM04B with Asterisk@Home 2.4 and Asterisk@Home 1.5 but I didn't find documentation about this installation. Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/85ff5252/attachment.htm

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

OK, I have a Asterisk @ Home 1.0.7 server with two Digium TDM400 cards, one 4 port FXO and one 4 port FXS. When I plug an analog cordless phone into the TDM40B card and setup a ZAP extension, the phone rings in and you can answer just fine. The weird part is when you try to get dialtone from the cordless handset it rings all of the other extensions... One other thing to mention is that the

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

Hi; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter

OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to

Hello all, I am having a serious problem installing my * with a TDM04B. I made everything work, call are coming in and going out including using a GSM Box in channel Zap/2-1. I did setup voicemail like this on extensions.conf: [incoming] exten => s,1,Dial(SIP/2246,20) exten => s,2,Wait,2 exten => s,3,Voicemail(u${ME}) exten => s,4,Hangup exten => s,102,Wait,2 exten =>

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

I am having a problem with a couple servers. They both have a Digium TE110P and a TDM04B card. I have setup the system so the TE110P uses channels 1-31 and the TDM04B 32-35. The problem is that when we reboot the server sometimes the TDM04B is recognized first and the TE110P second so the configuration fails. I do not know why this happens and to solve this I have to do a "service

I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The

Hi, A week or so ago I wrote about the problems I was having using a Digium TDM04B card in a Dell PowerEdge 750 IU running Fedora Core 1. Digium steadfastly indicates their card won't work in any PowerEdge 650, 700 and 750 series machines "do (sic) to a failure in interrupt handling between the pci bus and the card." My question to the list is: for those of you using a TDM04B

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can receive and place calls with no issues, however, when I receive a call, the CID only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone have any ideas even where to look to change this?? Is it a setting in the phone, Asterisk, or both?? Thanks, Marc

Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that