similar to: SIPAddHeader in .call file

Displaying 20 results from an estimated 9000 matches similar to: "SIPAddHeader in .call file"

2007 Oct 19
2
Live Conference about asterisk and voip: reminder 12:30 PM EDT Friday
As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference
2007 Dec 31
2
Problem with Polycom Soundpoint IP 320 Hardphone
Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able
2007 Jul 25
3
Asterisk 1.4.9.tar.gz download fails
Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E
2008 May 23
2
New York Asterisk Users
This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many
2007 Jul 25
1
WAV49 output in sox
Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.
2007 Jul 31
3
1and1 dedicated servers have been down for a few hours .
1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm
2007 Nov 01
1
AsteriskNOW and TDM800P
Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part
2007 Aug 29
1
OT - Callto:// tags options
Hello,
2007 Dec 17
1
dial, answered and then hangup
Hi all, I will a TDM card with FXO modules on it. Below is the dial plan. When someone can 9123456, CLI will show dialing to 123456 and answered. After answered, the call hangup. I would like to know what will cause the case to happen. Anyone can give me some advice to solve it? exten => _9X.,n,Dial(Zap/g0/${EXTEN:1}|${RINGTIMEOUT}) exten => _9X.,n,Hangup zapata.conf
2007 Dec 27
1
application not load
hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta
2007 Dec 27
1
Samsung iDCS 500R2 <PRI> Asterisk 1.4.*
I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS.... Anybody have any ideas? ________________________________________________________________ Sent via the WebMail system at
2007 Dec 11
1
Appending two voice files
Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one playback. TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Oct 15
2
Skills Based Routing
Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The
2008 Jan 02
2
Asterisk dialplan date and time operations
Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik
2007 Oct 23
2
text management
Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that
2007 Dec 14
3
GUI for Asterisk: Call Flow
Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.
2007 Jul 25
1
Add prefix digits in dialplan extention
Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so
2007 Jul 27
1
Asterisk Users Conference Friday at 12:30 PM EDT
You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http://x2z.eu for instructions on how to join or listen
2007 Jul 22
3
Debian etch and web voice mail - how to configure it?
Hi Everyone... I am running Asterisk 1.2.13 on Debian "Etch". I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says "It Works!" but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the
2007 Dec 27
3
Grandtream Conference issue
Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've