Re: [asterisk-users] a problem on Ubuntu with Asterisk

Scott Berry wrote: Have a nice day, Scott Berry E-mail: [EMAIL PROTECTED] I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I

Re: [asterisk-users] Differents routes for differents extensions

Create different contexts and assign them to the extensions [trunk1] exten = .X,1,Dial() [trunk2] exten = .X,Dial() and in sip.conf or iax.conf [exten1] ... context = trunk1 [exten2] context=trunk2 Marco escribió: Hi everybody, I need to use different outbound routes from

Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Matt Watson escribió: There is a .NET 1.1 library out there... I've played with it a little bit, but not enough that I could comment on how feature rich or stable it is... http://www.voip-info.org/wiki/view/Asterisk+.NET It'll more than likely not be compatible with AMI 1.1 however, which I

Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time

calllimit in sip.conf and you are done Vieri escribió: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use

Re: [asterisk-users] Send received fax to different email account

mark morreny escribió: Dear all, I am able to send and receive fax with Asterisk + iaxmodel + hylafax. What I want to be able to do is to 1. Stored the received fax in mysql 2. Send an email notification to he user corresponding to the incoming phone number 3. Send a SMS notification to

Re: [asterisk-users] Asterisk with lumenvox

Josué Conti escribió: Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josué ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Entering code to restart the machine

Steve Totaro escribió: On Wed, Feb 27, 2008 at 10:26 AM, Alex Balashov [EMAIL PROTECTED] wrote: bilal ghayyad wrote: Hi All; How can I configure Asterisk in that way: If I entered code (from my mobile when I call to the Asterisk or from any Internal Phone), then the machine do

Re: [asterisk-users] Restricting registration for peer 'iaxmodem0' to 60 seconds

Michelle Dupuis escribió: I have setup hylafax today, along with iaxmodem. I'm just starting the debugging process and see the following message every 60 seconds: [Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry: Restricting registration for peer 'iaxmodem0' to 60 seconds

Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)

Olivier wrote: 2007/6/15, Steve Underwood [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: ... The t38modem program from openh323 does this, and it has to do some nasty things to work. :-\ Steve Is this openh323 project alive ? Latest news date from 2003

Re: [asterisk-users] MeetMe Error

Manolet Gmail wrote: Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [EMAIL PROTECTED]:1] MeetMe(SIP/600-09111e58, 700|MI) in new stack

Re: [asterisk-users] MeetMe Error

Manolet Gmail wrote: 2007/4/18, Ronaldo [EMAIL PROTECTED]: Hi Manolet, You have to install zaptel in order to make MeetMe application to work. MeetMe needs a kind of timer device that is provided by zaptel package. Eventhough you don't have a zaptel card you need to install its package.

Re: [asterisk-users] a2billing

Pezhman Lali wrote: hi My a2billing adds |HrL automatically to dial string, I can not find the source of this task, I need to remove r from all dial strings, Thanks for your help. Best Mani __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the

[asterisk-users] transfers and CDR

Hi everybody, A question, how do I follow a call that is transferred? is the any event or something in the CDR that would let me find all the call sequence? Thanks Rodrigo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] h323 how to set it up?

Florea Igor wrote: Hi all, I have some questions about h323. Is it mandatory to install a oh323 or I can do h323 calls without patching or adding any new drivers ti asterisk? I did compile the asterisk with channel driver chan_h323 but it still gives me this error: [Feb 28 18:12:58]

Re: [asterisk-users] this i a test

Bayrouni wrote: Sorry for disturbing, but I sent some messages today and I am not seeing them on this list. Can sombody tell me, in case this message appear on the list. Thank you ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] jingle + asterisk 1.4

Hi, can someone give me a link to a howto about that? I want to use jabbin with asterisk but dont find how to register jabbin client in asterisk so it can make calls. Thanks Rodrigo ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Unable to launch Sendmail warning

Olivier wrote: Hi, From a bristuffed 1.0.8 Asterisk on a Gentoo system, I've got this : WARNING[8094]: Unable to launch '/usr/sbin/sendmail -t' Where could it come from ? Regards

Re: [asterisk-users] error when compiling zaptel-1.4

Alyed Tzompa wrote: The error lies here: make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. do you have the kernel-headers installed? (e.g. glibc-kernheaders-2.4-9.1.87.i386.rpm for Fedora) Alyed

Re: [asterisk-users] Can't get asterisk to compile chan_zap (was NewIssue)

David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes

Re: [asterisk-users] Can't get asterisk to compile chan_zap (was NewIssue)

David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes

Re: [asterisk-users] Asterisk very slow when internet down

Peter Mitchell wrote: Has anyone seen this issue with asterisk running like a dog when the internet is down ? Internal calls, incoming ISDN calls etc all seem to be affected. There is a local DNS server that is always available so I’m not sure why asterisk is so unresponsive. I’ve seen

Re: [asterisk-users] chan_sip loading delay in Asterisk 1.2.10

ast guy wrote: Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output -- Called SIP [EMAIL PROTECTED] and get ring back from B party... Is

Re: [asterisk-users] Re: Switching from FreeBSD to Linux - which distro?

yum can be used... direct download from http://isoredirect.centos.org/centos/4/os/i386/CentOS/RPMS/ Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so

Re: [asterisk-users] asterisk manager originate command

There is no dial command, I'm sending originate action from asterisk manager. Michael Collins wrote: I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Are you wanting the uniqueid for the call right after it is started, i.e.,

Re: [asterisk-users] asterisk manager originate command

My code is using phpagi-asmanagerbut what is sent is... Action: Originate Channel: SIP/802 Context: from-internal Exten: number to dial Priority: 1 Callerid: 802 Michael Collins wrote: There is no dial command, I'm sending originate action from asterisk manager. Oops, I

[asterisk-users] asterisk manager originate command

Hi everybody, I want to know how to get the uniqueid or a call started from asterisk manager using Originate command. Best regards Rodrigo Gonzalez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] app_prepaid won't load - undefined symbol mysql_num_fields

Install mysql devel package (depend on your distribution) and will work, it's not finding the library libmysqlclient Mosiuoa Tsietsi wrote: Hi again, I downloaded the bristuff-0.3.0-PRE-1s.tar.gz archive from http://www.junghanns.net which has a script you can run to download the sources

Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

User that web server is running has to have read permissions to file /var/spool/asterisk/voicemail/default/222/INBOX/msg.txt Easier option is to run apache as asterisk user Zeeshan Zakaria wrote: Anybody knows why ARI gives this error message when I enter extension number and password.

Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

If it is not parsing the index.php mean that you see the code in your browser, install php Alok Mohapatra wrote: Hi All, I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. After installing properly when opening in

Re: [asterisk-users] Something is trashing /var/run

mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed

Re: [asterisk-users] Lumenvox speech recognition

I've worked with it using Asterisk, and worked really fine Michael Welter wrote: Does anyone have experience with this product? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options