similar to: Sipura 2000 x dual g729 channels x other choices?

I have a Uniden wireless phone connected into Linksys/Supura 2100. It works well, except I never see any caller ID information displayed on the phone. Is that a setting in the 2100 that I'm missing, or is it an Asterisk setting or isn't it possible? Thanks, Jim.

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.

Hello All, I always used a compiled version for a x86 system >From http://kvin.lv/pub/Linux/Asterisk/ , and Works fine. At this time , I'm using a Pentium IV Dual core, Running Centos 4.3. I tried to install the 64 bits Compiled version but has a translation time > 20ms. Is it correct ? I also tried to make my self compilation and apply The patch but the patch doesn't work on ICC

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

Is anyone using a 3com 3CNJPSE to power a 7960G? I have a couple of 7960Gs and 3CNJPSEs but no combination appears to work. Both phones work fine with a cisco power cube. I get a 47.6V reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE. The network still works through the 3CNJPSE, just no power to the phone until I connect the power cube. Using two straight through wired

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

Hi everyone, I'm trying to send a FAX with the following configuration: Analog FAX machine (OKI) <----->SPA21000<----->LAN<----->Asterisk<--------> PSTN I'm restricted to use passthru mode for faxing, instead of T.38 protocol, because the Asterisk box is running v1.2 and cannot be changed as it is in a heavy production environment. Anyway, it "should"

I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't.

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 is behind NAT. I did the first obvious thing and mapped ports 5060 and 10000 - 30000 to the local IP address of the SPA-1001. Tried numerous proxy settings, have all the NAT settings == yes. Registration seems to be happening; with sip debug on, I see

Hi, I'm no phone guy but I've been having quite a bit of fun messing with Asterisk. I just unwrapped my new Wildcard TDM400P (TDM40B) which is the new 4 port FXS PCI card. It has 4 RJ45 connectors and I'd like to hookup 4 regular phones which are RJ11. If I crip RJ45 connectors to the phone cords do I simple use the center two pins (pins 4 and 5)? Thanks, Randy Smith Tiger

want to get from SIP to ISDN or from SIP to SIP. I only have a ADSL connection that means 786kb/s downstream and 128kb/s upstream so i can max handle 2 sip calls at once. I want to have 25 Accounts because of the different numbers for the different phones. Good, i wanted to buy a ready built pc from ebay but now i think i will built a 19" rack case so i will built in a 1 - 1,5ghz intel

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

I''ve just started using CentOS as an alternative OS for some servers for a project. At the time 3.4 was the release of choice. I''m curious how long the CentOS project will release fixes and patched rpms for 3.4 before it would be necessary to migrate these machines to 4.x. I rather know in advance so I can plan accordingly and slowly migrate these over time. I do realize that 4.x

Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the