similar to: ser problem

Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without

I''m trying to get Puppet to run on ESX v3.0.1. Being that ESX doesn''t come with ruby, I installed v1.8.6 under /opt/ruby/ruby-1.8.6 and linked the bin to /usr/bin. facter installed and runs without issue. However, when I try to install puppet, I get: - Could not load openssl; cannot install Is this due to the way I installed ruby or something else? Thanks, Clif

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Hi List; In the extensions.conf file, at the [global] context, there is a variable configured as: CONSOLE=Console/dsp What does it mean that? What dsp mean and it is shortcut for what? How can I use the core to get some data about such thing ambiguous for me? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Be a

Hi list; While I am writing my configuration on the .conf files, I would like to know if I wrote the command in write syntax (form), there is not any way to check if I am writing correct or not (other than checking my documentation)? Also, is there any method for searching on specific topic about asterisk (a command details and usage), from my computer (like help and so on)? Regards Bilal

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Josh Coalson wrote: > > > --- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > > > > > Hi all, > > > > > > Is seeking working for OggFlac files? I keep on getting a > > > FLAC__STREAM_DECODER_SEEK_ERROR. > > > > yes, it should work fine. in

Hello, I am using rsync at Mac OS X for synchronizing pictures for our two offices. Unfortunatelly yesterday the script stops working. Here is little workarround. The script select every file from folder A and write it to PENDING-FILES file. Than RSYNC take from PENDING-FILES every line (file) and transfer to folder B on different machine. Unfortunately some Mac user created folder started with

I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk

I'm trying to install centos 5 on an old machine, a Micron with a pentium III and 250 megs of ram. I'm using the DVD image. The media check said the DVD was OK. Everything starts fine (both in graphics and text modes), untl I get to the custom partition table. The new system will go on hda, with the boot and root partition reformatted (see below). I laid it out to preserve the data on

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [cybercall at ip-208-109-177-212 ~]$ asterisk -vvvvvvc -bash: asterisk: command not found [cybercall at ip-208-109-177-212 ~]$ I checked modules and other configuration files which are installed correctly. Please help me

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for