[Asterisk-Users] error while trying to install astcc

Hello list, Here is the error i’m getting when i try to « make install » with astcc. Can somebody know this error and how to fix it?   [EMAIL PROTECTED] astcc]# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi c

[Asterisk-Users] How to use ASTCC with SIP ??

Hello List,   I’ve set up asterisk and install astcc application, everything was well installed, but i’m having problem using astcc with SIP. I don’t have any Trunk card or any other analogic VoIP card connected to my asterisk box. I’m using SIP and asterisk-oh323 to connect to my VoIP pr

RE: [Asterisk-Users] How to use ASTCC with SIP ??

h file or directory == Spawn extension (prepaid, 77, 2) exited non-zero on 'SIP/8000104-71a3' Can somebody tell me why and how to solve it ?? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: samedi 29 janvie

[Asterisk-Users] Telephone Line options in Asterisk

Hello, Will asterisk support call waiting, call forwarding, CID, and three way calling if connected to a standard phone line? Or do I need to order these options from the phone company in order for asterisks to utilize them?. Basically I was wondering if I can order just a basic line for incom

[Asterisk-Users] Busy Extension Ring to alternate.

I know that with Voicemail you can either do voicemail(u) or voicemail(b), but with the Sipura SPA-841's I need to be able to roll lines from one extension to an alternate on the phone. For example: If extension 100 is busy, it will ring extension 120 on the same phone, and if that is busy it w

[Asterisk-Users] Individual contexts pending on Caller-ID?

Hi! Is it possible to handle incoming calls with different contexts pending on the callerid ? E.g. like you are able to define different contexts on each Zap-channel. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.d

[Asterisk-Users] Concurrent calls

Is there any way to quickly poll an asterisk server for concurrent call count? Preferably from like a perl or PHP script. -Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Re: Concurrent calls

:) On Thu, 3 Feb 2005 10:41:37 -0500, Daniel Corbe <[EMAIL PROTECTED]> wrote: > Is there any way to quickly poll an asterisk server for concurrent > call count? Preferably from like a perl or PHP script. > > -Daniel > ___ Asteri

[Asterisk-Users] ASTCC Apllication

the destination number. When I enter a destination number, the system says it’s not a recognized number and the call doesn’t go through. Can any one help me out with this issue? Is there a file where I can define extensions like in extensions.conf?   Thanks.   Daniel

[Asterisk-Users] How to Create customized audio file to use with ASTCC??

Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn’t match my needs (currency, etc.). is there any way to create my own audios and replace the current one??   Thanks.   Daniel

RE: [Asterisk-Users] ASTCC Apllication

Thanks a lot. Now I understand and it's working. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz Sent: vendredi 4 février 2005 15:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-

[Asterisk-Users] Which version of asterisk-oh323 should i use with asterisk v1-0-5.

Hi list, I have successfully upgrade my Asterisk V1-0-RC2 to V1-0-5, but I have a problem. The Asterisk box crashes now every time. I’m using asterisk-oh323. is there a stable version of asterisk-oh323 that can work with the v1-0-5 of Asterisk.   Thanks.       __

RE: [Asterisk-Users] How to Create customized audio file to use withASTCC??

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

RE: [Asterisk-Users] How to Create customized audio file to use with ASTCC??

ual wav >> files and then, finally, converted the sounds into gsm files. These >> sounds are being used in a low cost call shop in Dublin now. I'm not >> sure if my ASTCC recordings would suit your (or anyones) needs but if >> you would like a copy I have no problem pro

RE: [Asterisk-Users] How to Create customized audio file to use withASTCC??

convert them in gsm format. Thanks. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: lundi 7 février 2005 12:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to Create customiz

Re: [Asterisk-Users] TDM-400P and Grandstream Question

Hough, Linden wrote: Yes, I have a GS102 and you just plug it into your hub. It will get an address from DHCP if you have DHCP set up on your LAN. You can then get the address off the phone by hitting MENU - Down Arrow - Menu after it boots. You can then put the address into a web browser and confi

Re: [Asterisk-Users] TDM-400P and Grandstream Question

You can use rj11 jacks in the rj45 jacks. They will plug write in. Or use an rj45 connector with cat5 and route the other end to a punch down block or patch panel. When using an RJ45, the 2 center pins are the ones that will be used. Dan [EMAIL PROTECTED] wrote: Also, the TDM400P card will allo

[Asterisk-Users] DTMF CLIP in Sweden and others

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Are this currently working with CVS-HEAD? I've got an X100P-clone, and I've patched the zaptel drivers. But the Asterisk patches seems to be there. But I can't make it receive Caller-ID! Btw, by doing a cvs checkout asterisk, the HEAD-version will be do

[Asterisk-Users] Codec negotiation problems

frame type 256, while native formats is 4 (read/write = 4/4) Regards, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digiu

[Asterisk-Users] g729

ype 256, while native formats is 4 (read/write = 4/4) Now both channels in question have allow=ulaw and allow=g729 Any help at all would be appriciated. Regards, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

[Asterisk-Users] Re: g729

379968]: chan_sip.c:1797 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Feb 8 22:50:06 WARNING[1234379968]: chan_sip.c:1797 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Feb 8 22:50:06 WARNING[1234379968]: chan

RE: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9

. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: mercredi 9 février 2005 15:37 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9 > I get the following error when trying

RE: [Asterisk-Users] Web based Asterisk management tool

How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] G.729 codec for X-lite soft phone

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it?   Regards.   Daniel.   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

RE: [Asterisk-Users] G.729 codec for X-lite soft phone

, please simply ignore it.   Regards.   Daniel.   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: mercredi 9 février 2005 18:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] G.729 codec for

Re: [Asterisk-Users] Startup Question

Anton Krall wrote: Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and re

Re: [Asterisk-Users] why asterisk is replying 404 Not Found

What does your extensions.conf file look like? http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Dan Kamran Ahmad wrote: [3000] type=friend dtmfmode=INFO insecure=yes canreinvite=no auth=plaintext host=dynamic allow=ulaw

Re: [Asterisk-Users] asterisk@home scary log

You can always set up ssh to use host keys. Here are two howto's on what else? How to set them up. http://www.securityfocus.com/infocus/1806 Part 1 http://www.securityfocus.com/infocus/1810 Part 2 Dan. Steven Critchfield wrote: On Thu, 2005-02-10 at 09:08 -0700, Colin Anderson wrote: The hac

Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question

Very rough numbers: iax-gsm consumes about 22kb/s, I see about 60kb/s g711 about 80kb/s on I see 155kb/s Is that normal? This is an IAX link to voicepulse. I see all these lower numbers posted around but fail to see that on my connections. Using G711, Its only possible to have one connection a

Re: [Asterisk-Users] Searchable Mailing Lists & NooB Question

Rich Adamson wrote: Looks like your numbers add the transmit and receive data rates together, which is not a realistic way to discuss bandwidth consumption. An IAX link consumes about 22kb/s (round it to 30kb/s, who cares) in the transmit direction, and another 22kb/s in the receive direction. (The

RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.

I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED

[Asterisk-Users] Asterisk@Home 0.5

Hello list, Just wonder if [EMAIL PROTECTED] can work with asterisk-oh323 0.6. Did any one try it ??   Regards     ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users T

[Asterisk-Users] E1 and/or Euro-ISDN specifications?

Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

Re: [Asterisk-Users] E1 and/or Euro-ISDN specifications?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What's exactly Euro-ISDN? Is it G.931? I don't really get this. Is there a Q/G/E document for Euro-ISDN? I've downloaded two out of three fron ITU, so I would like to know for sure! :) Thanks! Peter Svensson wrote: | On Tue, 15 Feb 2005,

[Asterisk-Users] E&M and other Radio-based signalling

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Have anyone tried using this? I've looked at app_rpt, and that's a nice project, but have anyone tried using Asterisk for radio services using a Mux e.g.? I was thinking of using an E&M Mux (or channel bank i think) with TX/RX/BUSY/PTT functionality. Or

Re: [Asterisk-Users] Sangoma A104 - D-Channel problem

Hi, Did you tried to set your DMA or SATA as described at message "Sangoma A102 cards testing FIXED"? Daniel Kumak wrote: Hello, I have following problem with Sangoma A104 card: CLI> pri show span 1 Primary D-channel: 16 Status: Provisioned, Down, Active Switchtype: EuroISDN Typ

[Asterisk-Users] * Call Monitoring

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -D

Re: [Asterisk-Users] * Call Monitoring

Okay here's a quick and dirty little perl script to monitor the PRI Status and mimic nagios plugin output. -Daniel On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy <[EMAIL PROTECTED]> wrote: > On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe > <[EMAIL PROTECTED]> wrote:

[Asterisk-Users] Adit 600 MGCP configuration

I've finally got my Adit 600 and are configuring it right now. But I have to say, there aren't much documentation for it. I've setup MGCP and Asterisk seems to find it. But all channels (40 FXS channels) are "Down"! But the MGCP itself is "Up" according to the statistics. I can't find any documents

Re: [Asterisk-Users] * Call Monitoring

Yeah, I'd be interested in porting your work so it runs under nagios. Please post your results when you're finished. -Daniel On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev <[EMAIL PROTECTED]> wrote: > On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: > >

[Asterisk-Users] Amphenol cables?

A little off-topic maybe, but it's still for the Adit used with Asterisk. ;) I wonder where I can buy 50 pin Amphenol cables, with connector on one side, and open cables on the other for mounting in our own patch panels. In Europe, or Sweden preferably. It's said to be very common on telcos, but

[Asterisk-Users] More 2-way radio controlling in *

Audio out, PTT, Audio IN, Busy. 6-wire connection i guess? That should be a really nice setup with Asterisk! Anyone tried something like this? - -- Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCHsYS/

[Asterisk-Users] Asterisk in front of Toshiba CTX

I have googled, and wiki’ed until blue. Is it possible to put T1---*Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf   [channels] switchtype=national context=from-pstn

Re: [Asterisk-Users] Junk at the beginning, Warning, flexibel rate not heavily tested!

lie ka skrev: *Junk at the beginning Warning, flexibel rate not heavily tested! * Do not use MP3's with VBR (variable bitrate). Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-

Re: [Asterisk-Users] Warning[3817] and REGISTER

IN your Asterisk? Disable the chan_oss in modules.conf noload => chan_oss.so Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster

RE: [Asterisk-Users] VoipSupply.com

at voipsupply.com currently has the IP-600 on backorder. I placed an order for a few of them last week and they have yet to ship. I find it a little annoying that they don't inform you until after the order is placed that they don't have the item in st

[Asterisk-Users] AS5300 -> Meridian Configuration

DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 36 RCAP ND2 T200 3 T203 10 N200 3 N201 260 K7 Thanks, Aaron Daniel Senior Voice Analyst Sam Houston State University ___ Asterisk-Users mailing list Aste

Re: [Asterisk-Users] LOOKING TO HIRE

bullshit about things you don't know anything about from now on. Perl, PHP and Python are scripting languages. Daniel. smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] no music on hold

x27;s (CBR preferably) in your moh directory? Have you defined any moh classes in musiconhold.conf? Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/ma

Re: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc

ream firmware. Anyone noticed that they removed the BETA firmware from their site? Wonder why. Daniel (CISSP) smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium

Re: [Asterisk-Users] Voicemail wav49 format problem

Filepermission error or the mailbox doesn't exist. Check if /cygdrive/e/pbx/voicemail exists and it has the right permissions. (running under cygwin? cheesus..) Daniel Michael Stahl skrev: I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the

RE: [Asterisk-Users] Asterisk Project Consultant/Parner Wanted

Dear sir, I'm interested in your project. Can you tell more about it?? Regards. Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, May 22, 2005 4:23 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-

[Asterisk-Users] BudgeTone 101 doesn't register with FirmWare 1.5.23

he SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

[Asterisk-Users] DIAL with FastAGI and Answer Supervision

Hi! I'm using FastAGI (agi://) to make some calls. To do the dialing i use "EXEC DIAL Zap/g1/...". But how can I make "answer supervision" with FastAGI? DIAL command won't return until call is finished. Thanks in advance! -- Daniel __

Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare 1.5.23

Bob Goddard a écrit : On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0

[Asterisk-Users] VoIP-Forum.se - new Swedish user forum

, Daniel Nylander, VoIP-Forum.se CISSP smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

Re: [Asterisk-Users] Budgetone 102 and voicemail problem

rks for me. I'm running firmware 1.0.6.3 Regards, Daniel Nylander CISSP smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To U

[Asterisk-Users] SER Config for Asterisk

with this issue?? Thanks Daniel. <>___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SER Config For Asterisk

with this issue?? Thanks Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SER Config For Asterisk

meaning. Thanks. From: [EMAIL PROTECTED] on behalf of Arnd Vehling Sent: Thu 26/05/2005 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SER Config For Asterisk Daniel Eboa wrote: > This is the scenario i w

Re: [Asterisk-Users] SIP SoftPhone for debuging

direction. Why not use sipsak? http://sipsak.org/ Regards, Daniel smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or

[Asterisk-Users] H323 setup problem

hello. i'm new with asterisk but so far managed to make it work with SIP, IAX and even with 2 asterisk servers. Now i tried to configure it for H323 but it didn't work. i used net-meeting from microsoft. when i try to call from h323 to sip asterisk didn't say anything. probably the h323 client

[Asterisk-Users] Serious ZapRAS problem!

I have to reboot the whole server! The robot-voice is only on our side, it sounds fine at the other end. My server is an Dell PowerEdge 1850 with an Digium TE110P card to PRI. I'm using "stable" Asterisk 1.0.6 and I'm located in Sweden with

[Asterisk-Users] ISDN RAS and data calls

will reply with PPP directly. The both methods uses the same modem pool number. How can I tell Asterisk to inititate a data-call instead of voice-call? I'm using .call-files to connect to the ISP. -- Daniel ___ Asterisk-Users mailing list Asterisk-Us

[Asterisk-Users] .call files in outgoing dont get run

diffrent ways, but nothing works. What could be wrong? daniel eriksson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Digium PCI-e Cards

dardize on, and with a PCI-e card, that would be a lot easier. Aaron Daniel Sr. Voice Analyst Sam Houston State University ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSU

[Asterisk-Users] user web interface

Hi, I am using asterisk strictly as a voicemail server.  I know there are a number of web interfaces available-- I have looked at couple like AMP, etc...  Is there one in particular that is generally considered better than the others?  Is there one that is most feature rich with regard to managin

[Asterisk-Users] Problem with monitor.

Hi all, I got a small problem, my monitor is wacked :/ this is the result :( No clue what to send with from logs/etc since ther eis no debug info about it. http://www.1av10.nu/~puppet/auto-1118970177-0850007222-1002.mp3 -- Daniel Eriksson [EMAIL PROTECTED

[Asterisk-Users] Phone lookup

ld be appreciated. Aaron Daniel Sr. Voice Analyst Sam Houston State University ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] PPPD problem please help

from their side. The EchoReq's doesn't seems to reach them though. But the Terminal Request does afterwards. I think this is a Nobel Prize issue. :) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

[Asterisk-Users] ZapRAS

this can happen?! I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late though) not to work very well, but should that really be the problem with ZapRAS?! -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com htt

[Asterisk-Users] FOP related questions

, ...) either with FireFox and Internet Explorer. Any Idea? Best regards, Daniel ANDRÉ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Re: ZapRAS

>Daniel, > we have the same problem when our PRI line drops and Zapras has to >reconnect. You will also notice that the pppd process does not die >when Zapras does and the ppp connection cannot re-establish itself. >What we normally do is restart asterisk and then kill the ppp

[Asterisk-Users] Least Cost Routing

Hello, I am searching for a working solution for Least Cost Routing usable in France with asterisk. Does Anyone have any tip? Regards, Daniel ANDRE ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman

Re: [Asterisk-Users] flash panel only works with IP address

Carlos Chavez a écrit : On Thu, 2005-06-23 at 08:54 +0200, [EMAIL PROTECTED] wrote: ... There is a specific list for FOP you should directo your questions to. What is this list and how to subscribe to? Regards, Daniel ___ Asterisk

Re: [Asterisk-Users] FOP related questions

Seth Remington a écrit : On Wed, 2005-06-22 at 15:58 +0200, Daniel ANDRE wrote: Hello, I have downloaded and installed Flash Operator Panel. version 0.21. It works pretty well and I have some questions about it. 1. The text label of the buttons are partially hidden by their icons. Is

Re: [Asterisk-Users] Extension Matching.

My suggestion would be to basically strip the first digits down until you get to the 7 digit extension. Unless someone else has a better idea, that's how we do it, storing the first part in a variable in case we need it. Aaron Daniel Chris Modesitt wrote: Is there a way to match the l

[Asterisk-Users] Asterisk ---Toshiba

I set up a TE405P to go T1---*---Toshiba. I have the channels configured, and can place calls from the Toshiba, through * to the t1. Incoming calls work great to *, but if they go to the Toshiba, I get a hangup. I think the * is sending the call to the wrong span. I have 2 spans, span 1 from the T

[Asterisk-Users] 7905 example configs

Anyone have the example.txt file that Cisco's documentation on the 7905 IP phone keeps referring to? Or can someone possibly share a fairly complete example config file for these phones with me? Thanks! -Daniel ___ Asterisk-Users mailing list Ast

Re: [Asterisk-Users] No ringback over IAX - LiveVoip

On Fri, Mar 11, 2005 at 11:50:01PM +, Jay Milk wrote: > Dude, where have you been? This has been discussed here at length. > Everyone agrees that it's on LiveVOIP's end, but they're shrugging their > shoulders and pointing toward *. Search the list. Could you point out the best way to "sear

[Asterisk-Users] Realtime ODBC with cdr_odbc using the same database

ent dsn's in to connect to one database (i.e. [cdr-asterisk] for the cdr stuff, and [asterisk] for realtime) asterisk fails at loading the cdr_odbc driver. unixODBC doesn't throw any errors in its logs for that. Any ideas would be appreciated

[Asterisk-Users] session border control

has session border control been added to asterisk yet? i remember hearing about it, but i haven't been able to find any information on it on wiki. Thanks, daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

[Asterisk-Users] Redhat 9 Music on hold

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no in

RE: [Asterisk-Users] Pattern matching in extensions.conf

What is 00 and other numbers? Are different destinations prefix ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005 12:39 To: Asterisk maillist (asterisk-users@lists.digium.com) Subject: [Asterisk-Users] Pattern

[Asterisk-Users] Jabber module for asterisk

Would anyone know whether a jabber module would be in development for asterisk? What I'm looking for is something like the SER module that's out there, with capabilities to send SMS messages from jabber to a phone connected to the system. Aaron Daniel SHSU Computer Services [EMAIL

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 150

terisk Users Mailing List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed Jason Becker wrote: > Daniel Burget wrote: > >> I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 150

g List - Non-Commercial Discussion Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed Jason Becker wrote: > Daniel Burget wrote: > >> I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines >> connected via

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 186

It is not all that hard to do. Once you have all your phones setup with their own extension. When a call comes in from say 6000 Exten => 6000,1,Dial(SIP/phone1&SIP/phone2&SIP/phone3,20) Exten => 6000,2,Voicemail(vm#) Exten => 6000,3,Hangup This would ring SIP phones 1 through 3 all at the same t

[Asterisk-Users] MGCP issue

Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bin

[Asterisk-Users] Monitor command full static

I have a T1 going into *, SIP phones Grandstream & Polycom IP500. Everything works great, but when I use the monitor command, or use IP Switchboard to record a call, the call has really loud static, and you can only make out maybe 1 or 2 words spoken. I have tried the IN-OUT, and combined wav files

[Asterisk-Users] Monitor Command

Has anyone had any problems using the Monitor command? I get nothing but static. Zapbarge works fine, and I am using that, and recording calls, but if I could automate it, it would make my life much easier. Thanks. ___ Asterisk-Users mailing list Aster

[Asterisk-Users] Multiple CDR Locations

Does anyone know of a way to have asterisk save multiple cdr records in different places (i.e. the same record in a database locally and in another database on another system, or database and csv, or some other strange combination)? Aaron Daniel

Re: [Asterisk-Users] Multiple CDR Locations

CTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristof Hardy Sent: Tuesday, April 05, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple CDR Locations Aaron Daniel wrote: Does anyone know of a way to have asterisk save multiple cdr records in differ

Re: [Asterisk-Users] Digium Wildcard T1 Compatibility

Thank you everyone for your responses. Pretty much we're just looking for a way to bring in some lines with room for expansion to an aserisk server. We thought a T1 with a wildcard would be a good route since we could just have channels turned on as necessary. From everyone's comments it's soun

[Asterisk-Users] Problem with asterisk-oh323

Hello to all, I’m trying to get h323 working with Asterisk, I’ve downloaded all require modules (most are .tar.gz files, but if some body knows where to find working rpm file, it will help me), I’ve installed Pwlib, Openh323, and Asterisk. When I want to compile the asterisk-oh323 module,

[Asterisk-Users] G.729 for Asterisk: new version released

Download site: http://www.readytechnology.co.uk/open/g729 Major enhancements: - accepts packets with VAD stuff at the end (please test this with your hardware and give me feedback if it still doesn't work with some devices, you will probably see error messages on the console if bad sized frame

[Asterisk-Users] Faxing issues (no VoIP involved)

nyone have any pointers? Thanks, -- Daniel Jimenez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Strange error

Hello all,   I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All installation and packages compilation was successful. I have a SIP account to a SIP provider and I use it for outgoing calls. I’m using Cisco ATA boxes both SIP and H323, and al

RE: [Asterisk-Users] Strange error

Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Strange error you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa <[EMAIL PROTECTED

[Asterisk-Users] FW: Strange error

This is another error for the same config below : Nov 12 19:02:44 WARNING[1170207680]: chan_sip.c:1838 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 4/8)       From: Daniel Eboa Sent: vendredi 12 novembre 2004 18:16 To: 'Ast

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