similar to: IP Phones ?

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

Which router wichi support FXo ports do you recommend for -ASterisk SIP PBx? The smallest and cheapest? --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/6ec0ad0b/attachment.htm

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

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Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

What is the proper method to install/configure an X100P FXO card? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031009/ce8a7803/attachment.htm

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

I'm just wondering if anyone knows the story with these... http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=5721202362&ssPageName=STRK:MEWA:IT He claims they support IAX2 and SIP... but almost no history on the account selling them. I didn't see anything in the wiki about this company either.. Does anyone have any history with these phones? Thanks, Jared

Hi, Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible ? Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some

Hi, I have got one X100P telephone card in my Asterisk server and it's working well. I have two phones lines and would like to install a second card so I can use both lines. I have installed the card and tried to set it up, but all to no avail ! Could someone outline the changes that I need to make (and in which .conf files) in order to get the second card going ? Thanks in advance,

Hi, When I installed my X100P card there was a two ring delay before the phone started to ring. I managed to solve that by stopping the Caller Line ID being passed to the phone. I'm in the UK so that doesn't matter anyway. The problem that I have now is that when a call comes in on the external phone line and it is answered by a standard phone (not connected to Asterisk), the IP

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was

Hi there, I''ve read the paper on Xen live migration, and it shows some very impressive figures, like 165ms downtime on a running web server, and 50ms for a quake3 server. I installed CentOS 5 on 2 servers, each with 2x Xeon E5335 (quad-core), 2x Intel 80003ES2LAN Gb NICs. Then I installed 2 DomUs, also with CentOS 5. One NIC is connected to the LAN (on the same switch and VLAN), the

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

Hi there, I got really tired of false hangups, specially when someone calls from SIP/IAX/whatever to PSTN, which makes no sense in using busydetect in the zap channel, as the caller will eventually hear the busy tones and hangup, causing the zap channel to be "freed" as well... After suggesting that the busy detect be enabled only for inbound calls, and finding other people that think

Hi, I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have tried every conbination of T and t in the extensions.conf file, but all to no availe ! Can anyone help ? Thanks, Paul.

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server,

Hi, methinks that in the good 3 months since i ordered an IAXy, things have changed so much that now almost anybody out there with a VoIP hardweare website offers complete phones for less money than the IAXy, with support for both IAX2 and SIP in many cases, and fully configurable via its own keypad. What would you recommend as a rugged, small, and easy to configure/use self-contained unit that

I had this idea, and after looking for something like this already in progress, I found another guy who tried to start it... But I was unable to contact him, and his project seems to be dead. But, I believe it is possible, and I wanted to know the opinion of the experienced... So, let's go: I got an asterisk server setup to receive free calls from US to Brazil. The problem is that at my work,