similar to: ATA with IAX protocol

For those of you who wanted to know how the AG-468 (4xFXS) unit work (or it doesn't work), here is my personal experience. I had a problem from the start. The units ship from the factory set to static IP 192.168.1.200 even though it has a DHCP option in LAN Setting. So if you are on a different sub-net you have to figure out how to set your sub-net to access the unit; and those with less

Hello We have setup a doorbell which has an inbuilt analog phone which is connected to our Asterisk via a SPA2000 ATA. The problem we are getting is that when a caller presses the buzzer it is taking two or more minutes to finally call the reception phone. In the SPA2000 I have set dtmfmode to be inband. I notice that with the asterisk you dial a number and then it waits for a timeout

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

Hi Is there a preferred setup of IAX/2 with regards to silence suppression or are there any "hidden" little secrets that I should know about? I have a configuration that works pretty well, however there are at times "silence" on the phone and you are not sure if there person is still connected. I am using grandstream Budgetone-100 ip phones. Does this make any sort

Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via "make samples". Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

Hello, I'm trying to find a good solution for load-balancing of several Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have a front VoIP router (or several) who dispatches the calls of the different boxes. This routing can be done with SER for SIP (redirect server) & GnuGK for H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk box to act as

Guys. Maybe this is asking for a lot :) but is there any web client that can use IAX2 and ilbc? This is for a "call us" web idea.... Any leads?

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel? There is a new application from Newman Telecom for fax detection. http://www.sineapps.com/news.php?rssid=575 Current Asterisk Fax detection doesn't work for me as I don't have Digium cards; I'm using Siupra -- #Joseph

As the title suggests, I was wondering if there was another source of documentation for Asterisk. Related: If one wanted to contribute to documentation, who would one contact? Thanks! Sean

Hello In my extensions.conf file: [frompstnisdn] exten => s,1,Dial(SIP/200&SIP/202,20) exten => s,2,Voicemail(su200) exten => s,3,Hangup I use the s, start, extension to handle incoming calls. In my zapata.conf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn'

Hi, I'm looking for a tutorial or installation guide for SER to be used with asterisk to solve the remote SIP agent problem. All the documents available are for large scale installation. Any help is highly appreciated. Regards. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

I have an asterisk box and SIP / IAX2 phones. To call out, users have to add 0 (zero) before a real telephone number. That means, that if they want to call someone that has a number 123456, they have to call 0-123456. Simple, right? This has a serious drawback though - when someone calls us from the number 123456, we see the callerID 123456, and we're unable to use the callback/redial

Are there any IAX speaking "hardphones" out there? If so, can anyone offer comment on their quality? Thanks! -Dorn

Hi all! Somebody knows something to do with a high delay using Asterisk + DIAX!? When I used IAXComm(Linux) in both sides(peer and me) no problems. Whan I used DIAX099f(WinXP) in both sides(peer and me) I have a delay in the voice coming from the person that I called. I don't have delay in my voice to the peer phone. CODEC: u-law (I tried with all available codecs) Thanks for your help!

Hi Can TDM400P detect polarity reversal on FXO module? We have C.O. lines that reverse polarity on Answer and release. Thank You This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you