similar to: Linux IAX client

I have a UIP200 on the way for eval. Does anyone have tips or tricks to get it working right away with * ? I hate having to go through the pain someone else braver than I went through already. :) Tim --

It would be a real kick to get one of these to run Asterisk. :) http://www.gumstix.com/ Tim -- >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >>

What is the best inexpensive voip phone out there? I want to try a few with *, but don't want to go broke while I'm just playing around... Tim --

Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... Tim --

I have a system that has a TDM400 with 3 FXO modules. Recently, the system started not seeing the ports. No answer, no lines available for oubound dial, and only a reboot of the machine will bring it back. Any ideas to try? Thanks, Tim --

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've tested the S100U unit in two different computers both with different USB subsystems. The one it worked the best in used the usb-uhci USB driver. The other system uses the usb-ohci USB driver. To make sure it was not the installation, I used the same hard drive that had asterisk in both systems. And I moved the X101P and S100U between both

Folks, I'm trying to install one of the new quad fxo cards remotely. I know the existing machine was too old to have a PCI 2.2 bus, so I had my helper at the other end try a few boxes that were sitting on a shelf with the new card and a Knoppix cd. He found one that reported the card as the Tiger Jet. Good. Now, we moved the HD from the existing machine, loaded with Debian, and the card is

I've been wondering what the difference is in the syntax of things, like Dial. Some examples show things like: exten => 500,1,Dial,SIP/${EXTEN}|10 but other examples show: exten => 500,1,Dial(SIP/${EXTEN}|10) or exten => 500,1,Dial(SIP/${EXTEN},10) Which one is correct? Or most correct? Which one is preferred, and why? I'm sure I'm not the only one with this question... :)

Does anyone still have the 0.7.2 debs hanging around? I need to revert a recent upgrade. We're having too may flaky problems (like softphones being able to dial out fine, but GrandStreams failing to dial every other time), and iaxcomm not working with gsm. Thanks, Tim --

I have my asterisk box on the public network. I have a winders box on the public network, running diax. I have a winders box, same setup, behind my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native bridge, the call fails. I've poked at this way too much today... what ports do I need to open for

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've searched the mailing list quite extensively, but didn't come up with anything promising (some things wer helpful, though). Does anyone know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310) phones can be made to work with the TDM400P card or if they are ADSI compatible at all? I kind of doubt they will work if they are

Ok I have a question. I have Xten-lite working with our Asterisk system and I am able to make and get calls. But the main problem is the sound is very choppy and sometimes it cuts off words. I have tested it with ulaw and alaw as well as GSM. They all do the same. ulaw seems to work better. I also have an ATA-186 which works great without this problem. Here is my Sip.conf settings.

I am attempting to setup a simple network-to-network IPSec tunnel. The tunnel appears to be setup correctly because I can make connections between the networks and tcpdump shows esp packets going between the two gateways. My problem is that I cannot make connections from one gateway to the other through the tunnel. I think that this is a routing issue. Here is some more info about my network:

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

http://www.skype.net/download_pda.html http://australianit.news.com.au/articles/0,7204,9214802%5e16123%5e%5enbv %5e,00.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/51a98542/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, been awhile since I''ve written. I now have a situation where I get to use traffic shaping for a client. ~ We implemented the WonderShaper script on our own firewall and experienced no problems. I made some modifications to it to add IPSec protocol packets into the 1:10 high priority class using the u32 filter. ~ So far on our

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy

Can PDAs be used as softphones/clients on asterisk? what i wanted to do is to set up 2 PDAs as softphone(client) which allows them to communicate each other through asterisk server(desktop) devices i have: pda compaq model 3680 pda sharp sl5500 access point desktop(asterisk) can i apply my idea on the asterisk? any guides? thanks in advance :) --------------------------------- Do

I wanted to post the beginings of my latest IVR Project for an automated Time Clock software. The customer has over 300 Field Reps that call in everytime they arrive on location and whey they leave that location. This is handled by the receptionist now and she logs in them and out of there Time Clock Software. Which takes up majority of her day. The customer has requested a automated way of

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working