similar to: gumstix

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 --

Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, 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 --

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

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 --

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

Has there been any luck getting the 3Com NBX series phones to work with Asterisk? Thanks! -Andrew

Stutter tone when voicemail in boxHi, I'm having a problem getting Asterisk to make a stutter tone when a voicemail is received. It used to work, but since I rebuilt my server I can't get it going. I've set put a line in the sip.conf file: mailbox=100 to check it, but it's not doing the stutter tone ! I am using the stable version of asterisk from the CVS tree. Any help /

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 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 --

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

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

PROBLEM: My x100p only does bad things. When I plug the line from the wall into the card, other phones in my house go nuts, no dialtone, crazy clicking, random tones, etc. Following the steps below, the line is screwed up any time I test after step 2. Just plugging the line from the wall into the x100p takes my other phones off line. External calls to my line hear busy. Obviously, can't

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 --

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

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Hi, I want to build a PBX base on Asterisk using an embedded device. Can anyone please recommend an embedded device I can use for doing so? I will install linux or freebsd in the device. Thanks A

So, I just got my GumStix Stuart BT today: http://www.gumstix.com For a non-telephony (Bluetooth based) project. I'm browsing the SVN website for Gumstix and lo and behold, there is Asterisk! I'm excited. Has anyone ever tried it on a GumStix before, and if so, care to share tips? If anybody is curious, this is a PC the size of 2 packs of Wrigley's Spearmint Gum put together. Runs

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"