similar to: On Hold music

We've got some replies to questions online about Astricon and we now have a mirror available at: http://astricon.voctel.com/news.php If anyone has any comments about Astricon, please forward them to me and I will put them up on the site so that all the people who didn't go can read them. Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html)

Hi, Just thought I'd let everyone know of our latest interview - this time with Mark Spencer - the creator of Asterisk. -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

Hi, I've just done up a quick proggy to show me the status of my IAX peers from my windows box. It plugs into the simple manager proxy. You can see more information (including a screenshot) at: http://www.sineapps.com/news.php?rssid=384 You can download it directly from: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. -- Cheers,

OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:

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

Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Shane Breen > Sent: Wednesday,

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

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

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?

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'

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

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

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

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

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!

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

Please send me a quote for remote installation of Asterisk, GUI administration, and billing for calling card, caller ID based prepaid, and postpaid. Off list please. ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Sorry, I don't remember who was asking about this, but it seems that if you record a video message that contains the send video start, it will actually fire up the remote receive window. I.E. Previously I was using the recording section of voicemail to create my video IVR's. This meant that when I arrived at the section to record the message, I had already clicked send video in Xten.