similar to: Line is stuck off hook...

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

First question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B (I have 2 computers with the same setup), will I be able to use SIP? I already have the two computers calling each other using IAX, but I would like to see if I can get them to work using SIP. My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0

Hello All, I could use a recommendation if anyone has a moment. I have the T100P but I have not gotten my service yet. I want to have at least 12 lines of digital voice with DID. Should I just seek out a PRI ISDN provider or is there something else I should look for? I want to keep cost as low as possible. Also, I want to own my own router for the phones since it is always a hassle to get

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Has anyone on the list actually purchased any H3500CW from http://lktelecom.zoovy.com? I need to purchase quite a few phones and I was looking to get the H3500CW but it appears they are in China so I was wondering 1.) if they are a reputable company (before I put the $ out) 2.) about how long it took for shipping because I need these phones pretty quickly. Thanks all AJ

Hi, I've been browsing for FXO devices, and I'm really surprised at their costs. Why such devices are so expensive and somehow hard to get ? Samy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/a92ee327/attachment.htm

Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). > -----Original Message----- > From: John Schmerold [mailto:john@katy.com] > Sent: Tuesday, 5 August 2003 12:37 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Zhone Zplex 10 units > > Thanks for the Zplex heads up. > > Steven

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am!

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be

Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case

Hi 2 all. I've a nice issue my * system. I have: asterisk latest cvs T100P + Zhone cb (3fxo+10fxs used) Diva Server 4BRI running Capi a bunch of pots and sip phones The whole system is running good (thanks to capigod and Bic. for the help with the Eicon Diva), but there's a very strange issue when calls arrive on the Diva and then bridged to the pots on the channel bank: the voice is not

I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) to use at home only to find that the fan noise is too loud. As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port FXO card? Gene

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ?

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm