similar to: Problem with Manager Interface

Displaying 20 results from an estimated 10000 matches similar to: "Problem with Manager Interface"

2005 Mar 18
2
Parking a call in manager interface
Is it possible to park a call through the manager interface? If yes; how? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/ec2a5f90/attachment.htm
2005 Jan 08
1
No such extension {Scanned}
Hello All, I'm trying to dial out with no luck. I'm using Asterisk@Home defaults. I have one X100P card and SJPhone. *CLI> dial 96985628 No such extension '96985628' in context 'default' Here is my exten [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
2005 Sep 20
6
iax2 trunking wackyness
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
2005 Jan 17
3
On Hold music
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:
2005 May 16
4
Web Client with IAX2 and ilbc
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?
2004 Sep 21
12
Astricon pictures
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
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
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
2005 May 25
4
SER Help
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
2006 Jan 11
3
video development
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
2005 Mar 15
2
Wiki down: Is there another source for documentation?
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
2005 Sep 30
2
Why does the s extension not work in my extensions.conf file
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'
2005 Sep 30
2
analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly
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
2005 Jan 13
3
High delay with diax099f + Asterisk
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!
2005 May 25
5
Asterisk Crashing; Not getting Core dumps
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 --
2005 Jun 05
2
TDM400P Polarity reversal detection
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
2006 May 26
4
mpg123 or asterisk
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
2005 Aug 28
2
Need quote for Asterisk and billing remote install
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
2005 Aug 28
4
Mplayer as replacement to mgp123 in MP3Player cmd
There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of
2004 Dec 01
4
Voicemail - Danish, German an French audio files download?
Hi all, Is it possible to download Danish, German and French audio files for Asterisk somewhere, or does everybody just record them? Thank you in advance Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/02cec53d/attachment.htm
2005 May 23
4
Digium FXS modules too fragile?
Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.