[asterisk-users] sox gsm

sox needs for gsm an optional library. I was not able to locate this one. Can anybody point me to this place? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Re: Asterisk Jobs Update

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For more information or to start looking for Open source Asterisk VOIP employment head over to http://www.asterisk-jobs.com Is it that I don't know how to make search or there is no jobs available in any country? -- Tomislav Parčina

Re: [asterisk-users] Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)

Dear Leo,As you said, I have tried using dtmf and in different values. But, no reuslt. Finally, I knew that Basic Asterisk setup doesn't recognize callerid in India. To get callerid in India, we have to do some modifications in chan_zap.c source file. But, I dont know what modifications I have to

Re: [asterisk-users] sox gsm

On Mon, Aug 21, 2006 at 02:02:31PM +0800, Ronald Wiplinger wrote: sox needs for gsm an optional library. I was not able to locate this one. Can anybody point me to this place? As there is a Debian package you can grab the orig tarball from: http://packages.debian.org/unstable/libs/libgsm1

[asterisk-users] no audio issue

Hi im experiencing no audio problems. ive installed the latest asterisk 1.2.10 zaptel, libpri asterisk. the caller's side reception is fine but i hear nothing on my sip account. Please help Regards, John ___ --Bandwidth and Colocation provided by

[asterisk-users] queuememberstatus overwhelms manager socket connection to asterisk

I have a test application, what it does is just connect to the asterisk manager,and listen for events. I also set the connection to receive on user, call and agent events.I Noticed that everytime the queue is empty, asterisk tends to throw too many queuememberstatus events, overwhelming the

[asterisk-users] Re: no audio issue

there's actually no audio b/w sip to sip calls. I just tried 2 sip extensions and there was no audio in any of them. what could be wrong? nat = 1 i used ulaw and then gsm and one of them worked. im not using qualify. i have tried everything i could think of, even applied the patch

Re: [asterisk-users] Asterisk installations in Germany

asterisk-robert wrote: I need to send some information to our German HQ regarding my experiences with VoIP. Asterisk is very prominent in those experiences. I would like to include information about installations of Asterisk at German companies/universities. We have installed Asterisk

[asterisk-users] Is there an [EMAIL PROTECTED] specific list?

Thanks in advance Paul ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] IAX2 Auto fallthrough

Hi all,Could anyone help me, why my calls of some clients disconnecting with the following error message: i have more than 500 IAX users but it is happening with very few customers.I will be appricate for u kind of help.Error::Auto fallthrough, channel 'IAX2/2001@2001/1' status is 'UNKNOWN'

Re: [asterisk-users] Polycom 601 Issues

On 20/08/2006, at 8:38 PM, Paul Hales wrote: Does anything pop up on the Asterisk screen? Does music on hold work fine? PaulH On Fri, 2006-08-18 at 13:13 +0800, Nathan Alberti wrote: Nothing strange on the asterisk console... just stopped and started hold on channel. If I

[asterisk-users] SIP ActiveX?

Hi, I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi. Thanks Kind Regards, Lennie De Villiers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Re: no audio issue ([EMAIL PROTECTED])

U need 2 give more info on your setup i.e wherether u have sipclientasterisknatsipclient or whatever the situation is . Anyway in the mean time just rtp dubug on and see wherether there r rtp packets sent back and forth there's actually no audio b/w sip to sip calls. I just tried 2 sip

[asterisk-users] Re: Asterisk 'Hosting'

JM == Jeremy McNamara [EMAIL PROTECTED] writes: JM Why do you need multiple instances? Just setup your Asterisk JM configuration to separate the various 'customers' or 'tenants'. The configuration files balloon to unmanageable sizes, and changing them means that you risk breaking telephony for

[asterisk-users] zap channel media volume

Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk

[asterisk-users] Re: Asterisk 'Hosting'

MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes: MR And so you're thinking it would be better to run several hundred MR Asterisk instances?! Why not? As long as you stay away from the things that need zap timing, asterisk is really not much of a load. /Benny

Re: [asterisk-users] Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units

Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Marco, as good? Well, you are use libpri-1.2.3? Believe that this is a bug of this version. Look at link´s below, contains patchs for this problem. I wait to have helped. Best Regards Josué

Re: [asterisk-users] Ignoring PRI call?

So, a few questions: - If the call received by asterisk from the PRI is sent to a number not in the dialplan, what will asterisk do? Will the call be cancelled, or will asterisk signal something back to the switch to indicate dunno about this, try another? Asterisk will do whatever

Re: [asterisk-users] SIP ActiveX?

Lennie De Villiers wrote: Hi, I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi. Thanks You can find a proof of concept at http://www.pernau.at/kd/voip/bookmarks-sip-phones.html It's called ActXPhone regards klaus ___

[asterisk-users] Zaptel install - Fedora Core 5

I'm trying to install Zaptel 1.2.7 on Fedora Core 5 with 2.6.17-1.2174_FC5 kernel from source code. When I untar Zaptel and execute this is error that I get. cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE

Re: [asterisk-users] Zaptel install - Fedora Core 5

I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that kernel. You could probably create the devel package without

[asterisk-users] Re: Re: PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi! This patch does passthrough the AOC information from on ZAP channel to another ZAP channel. There is no support yet for storing the AOC value as CDR, but I think this may be easily added. Hi Klaus! I'm not programmer so I don't

RE: [asterisk-users] Is there an [EMAIL PROTECTED] specific list?

Yeh but as [EMAIL PROTECTED] is now called Trixbox so go to www.trixbox.com Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul A Brown Sent: Monday, 21 August 2006 3:58 AM To: Asterisk Users Mailing List -

Re: [asterisk-users] Announce caller-id

I did something along these lines, but I was playing the caller ID back to the caller, not after a transfer. In a perl AGI script. I split the caller ID number into an array, seperated by '//' so each number was an element. Then I played digits/$array[0]... digits/$array[1]...etc. coolbreeze

[asterisk-users] running agi application in the background

I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. When the call is answered, the application checks to see the number of seconds (talk

RE: [asterisk-users] Zaptel install - Fedora Core 5

I couldn't find 2.6.17-1 for download but this is what I used to install the kernel source http://download.fedora.redhat.com/pub/fedora/linux/core/5/source/SRPMS/GFS-kernel-2.6.15.1-5.FC5.17.src.rpm Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED]

[asterisk-users] Re: Zaptel install - Fedora Core 5

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I managed to get zaptel to compile reasonably easily on 2.6.17-1.2157_FC5smp. However, the yum repo sites do not provide devel packages for 2.6.17-2174 for some reason last time I checked, hence couldn't get it to build on that

[asterisk-users] how to set 'transfercapability'

hi, we've got a Diva Server BRI-2M PCI, SN:3485 card in our asterisk server. we use the latest divas4linux-melware-3.0.g-106.628.1-1 driver for it. the card is connected to a Bosch Integral33 PBX. the two system connected with an S0 line in order the two pbx be able to call each other. when we

Re: [asterisk-users] Re: Zaptel install - Fedora Core 5

In which case your best bet is probably to install with an rpm -- rebuilt on the source rpm. simon On 21 Aug 2006, at 12:36, Tomislav Parčina wrote: In article 344F8B3D-6591-4001-9DE6- [EMAIL PROTECTED], [EMAIL PROTECTED] says... I managed to get zaptel to compile reasonably easily on

[asterisk-users] polycom_acd_functions branch and outboundproxy

Hi, I'm using the polycom branch and have been trying to get the outboundproxy=xxx to work. Is this something that should work in the version of software? Thanks, Dean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] Ignoring PRI call?

On Sunday 20 August 2006 10:55, Roy Sigurd Karlsbakk wrote: - If the call received by asterisk from the PRI is sent to a number not in the dialplan, what will asterisk do? Will the call be cancelled, or will asterisk signal something back to the switch to indicate dunno about this, try

Re: [asterisk-users] running agi application in the background

Allan Kamau schrieb: I would like to run a fast-agi application in the background.(cmd agi()) This is because I would like to implement a disconnect after so many seconds feature or at least a log of the duration of the call. What about using an Option of the Dial-App instead ?? S(n):

Re: [asterisk-users] Sending signals to asterisk

On Sun, Aug 20, 2006 at 06:20:42PM -0300, Danko Miocevic wrote: Hello, is there any way to send signals to asterisk, for example, I send a sign to a parallel port and it calls an extension. I can´t modify asterisk code to make it. Any ideas? Thanks for your time,

[asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?

Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the

[asterisk-users] Joining calls via manager.api or AGI

Is there a way to initiate 2 different calls and connect them together with Asterisk, using the manager.api or the AGI system? I want to link the calls without using DTMF, such as with an SMS or web triggered script. I thought the call files would be able to set the necessary AGI variables for

Re: [asterisk-users] Joining calls via manager.api or AGI

Is there a way to initiate 2 different calls and connect them together with Asterisk, using the manager.api or the AGI system? I want to link the calls without using DTMF, such as with an SMS or web triggered script. The only way right now is using meetme. There is a patch with a 'bridge'

[asterisk-users] Configure mailserver to deliver voicemail

G'Day List, I am looking for documentation on how to configure sendmail to deliver asterisk voicemails to the recipient's mailbox. I Googled it but found many many references to the fact that asterisk can do that but no How-To's. I believe sendmail is running on my asterisk box as:

Re: [asterisk-users] Joining calls via manager.api or AGI

This feature was supposed to be in 1.2, in fact Kevin promised me that it would be since I had it in before the feature freeze for 1.2. It did not go in. Since then I have had to move on to other things and others have tried to keep it going. This is really a very basic function that should be in

[asterisk-users] IAX2 TRUNK CPU consumption

Hi, I have a strange problem about the cpu consumption of a IAX trunk. I have two asterisk connected by a IAX trunk. The asterisk number 1 is installed on a Soekris Box Asterisk 1 Asterisk 2 IAX T | -- | I use another asterisk to generate some traffic

[asterisk-users] Is it possible to call System dialplan application via AMI?

Hi guys, Does anyone know whether is it possible to call System (Execute a system Linux shell command) dialplan application via AMI? If so, how? Thanks in advance, * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] Configure mailserver to deliver voicemail

On Mon, Aug 21, 2006 at 09:52:23AM -0400, Ferguson, Michael wrote: G'Day List, I am looking for documentation on how to configure sendmail to deliver asterisk voicemails to the recipient's mailbox. Nothing special about sendmail. Basically any standard MTA: sendmail, postfix, or

[asterisk-users] t.38 bounty

hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks --- Marek Cervenka ===

SV: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?

Sorry. It sould say SIP-to-Zap not the other way around. Meaning that the Zap user is heard fine, but the external-SIP user is choppy when calling out on Zap (not when calling SIP-to-SIP though). -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL

Re: [asterisk-users] Apache for FastAGI

On 8/18/06, Shidan [EMAIL PROTECTED] wrote: I don't know if I responded to the original poster before but if you are looking for a python fastAGI server, there already is one, its called starpy. Anders, since you know Erlang, do you know of any media processig libraries in Erlang, do the

Re: [asterisk-users] Ignoring PRI call?

This is what I am trying to do, yes, as to do all DID administration myself without contacting the switch monkey. It's quite possible, it seems, by sending a cause 34, lying about no bchans being available to handle the call. Thanks for reporting back, I like this idea :) thanks again.

Re: [asterisk-users] Analog-to-VoIP: blade?

www.zhone.com. Their MALC can handle 500 POTS lines in a 23 shelf with POTS - VoIP (SIP/MGCP). 'Telco quality' and the per port cost for high density isn't that bad. You could probably also go with a bunch of CAC AccesBanks connected to a CAC Widebank, connected to a Lucent TNT and

RE: [asterisk-users] Re: Asterisk 'Hosting'

-Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Monday, August 21, 2006 3:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk 'Hosting' JM == Jeremy McNamara [EMAIL PROTECTED] writes: JM Why do you need multiple instances?

Re: [asterisk-users] SIP ActiveX?

This is a commercial activex you may want to evaluate:http:/.www.vaxvoip.comIt worksLennie De Villiers [EMAIL PROTECTED] wrote: Hi,I'm looking for a SIP ActiveX component to use in Visual Basic/Delphi.ThanksKind Regards,Lennie De Villiers

Re: [asterisk-users] t.38 bounty

marek cervenka wrote: hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks Do you really think T.38 can be implemented on a contract basis for $11,750? Besides,

[asterisk-users] DTMF + voipjet

Hello list, Was wondering if anyone knows how to get DTMF to work on voipjet.. Tried, dtmf=rfc2833 dtmfmode=rfc2833 doesn't seem to work... Any clues? Cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

[asterisk-users] Cancelling outbound call: is Asterisk behaving correctly

Hi, we have a setup with an Asterisk, an openser and a Cisco 5400 in place. Asterisk is the frontend to the users, providing registering and RTP proxy functionality and openser is the gate-keeper of the Cisco. I can call in and out, everything is fine so far. But there is one strange fact:

[asterisk-users] failed calls

I am trying to track down a problem which is occurring on about 1% of the phone calls through a customers system. Layout looks like this: PSTN PRI Asterisk A IAX Trunk over point to point T1 Asterisk B SIP over LAN Polycom IP501 1) The user on the Polycom IP501 phone dials

[asterisk-users] Size of realtime appdata field under MySQL

Hi all, I'm trying to use a bigger appdata column for realtime, the reason being that I'm moving to a new setup where the SIP devices are named according to the name of the user and some of my dial/page commands need to dial a goodly number of phones which then exceeds the 255 max size of the

[asterisk-users] Asterisk in Xen 3.0

Hello! Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? Regards, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com --

RE: [asterisk-users] Announce caller-id

How where you able to interact with the callee after they had answered the call? You lose control of the dial plan after someone answers, until they hang up. -Original Message- From: Roy Kidder [mailto:[EMAIL PROTECTED] Sent: Monday, August 21, 2006 5:05 AM To: Asterisk Users

[asterisk-users] Status of Monitor

Is there a way to find out if a channel is currently being recorded/monitored via the Asterisk Manager API. Currently, if I issue a Action: Status, it lists all channels as unmonitored, regardless if they're being recorded or not. (In my setup, I'm not doing automatic monitoring, I have a

Re: [asterisk-users] Asterisk in Xen 3.0

Hi, Tomer Horn wrote: Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? We do this a lot, although I believe our engineers are still using Xen2 for systems with BRI/PRI

Re: [asterisk-users] Polycom IP430 won't finish boot

Hi Doug - Let me start by saying when I first plugged it in, I didn't have the files set up on my ftp server yet, and the phone used it's default settings and it completed bootup. Now... I started with sip v1.6.6b and bootrom 3.1.3 on the ftp server. Phone boots, d/l's files, reaches Welcome

[asterisk-users] Portuguese sound files available?

Hi, I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Regards, Ricardo.

[asterisk-users] Variable to show caller id for a current call?

Is there a variable that can be gotten with GetVar to show the callerid of the current incoming call in progress at a sip extension? For instance, a caller from 516-922-9463 calls extension 234. I would like to be able to be able to get back the 516-922-9463 if I pass 234. Also, can this be

Re: [asterisk-users] t.38 bounty

Steve Underwood wrote: marek cervenka wrote: hi, bounty for t.38 is $11,750. that looks good! http://www.voip-info.org/wiki/view/Asterisk+T.38+Bounty how high must be bounty for Digium to hire programmer for this? thanks Do you really think T.38 can be implemented on a contract basis

Re: [asterisk-users] t.38 bounty

What is the status of it anyway? I followed the bug for it and it appears that the bug was closed and maybe it was incorporated into Trunk. Is this true? And should it be (fully) functional now? PA ___ --Bandwidth and Colocation provided by

[asterisk-users] Re: Zapand SendDTMF??

I have tried it with exten = 5481,2,DIAL(IAX2/5480,,D(w1)) and it also does not work. I have since moved it to an analog extension on a legacy PBX. I have tried: exten = 5481,3,DIAL(Zap/g2/5110,,D(1)) and a macro with SendDTMF. It works fine if I dial 5110, then enter the number of the

RE: [asterisk-users] Variable to show caller id for a current call?

${CALLERID(number)} -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: Monday, August 21, 2006 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Variable to show caller id for a

[asterisk-users] Realtime and labels

Does anyone know if realtime extensions support the use of labels? ie: exten = acdpause,1,Answer exten = acdpause,n,Wait,1 exten = acdpause,n,PauseQueueMember(|Agent/${CALLERIDNUM}) exten = acdpause,n,GotoIf($[${PQMSTATUS} =

Re: [asterisk-users] Metermaid - Parking Slot

David Gagnon wrote: Finally, in the trunk all the states of my device are broken. If I downgrade to 1.2.10, everything is fine. The device get busy and ringing. But in the current trunk Asterisk SVN-trunk-r40632M none of my hints works. Anyone could confim this bugs ? David, I haven't

[asterisk-users] Realtime and hints

Can realtime be used with hints? How would you get the following into the database given that the priority column is numeric, and that there is no application for the first entry? exten = 2944006,hint,SIP/2944006 exten = 2944006,1,Dial(SIP/2944006) Every time I touch realtime I hit obstacles.

Re: [asterisk-users] Variable to show caller id for a current call?

But how do you get that with GetVar? I am trying to do this through the API. I tried: Action: GetVar Variable CALLERID(227) and I tries: Action: GetVar Variable ${CALLERID(227)} Neither returned anything. How can I do this? Alternately... Is there a way to have a program fired off when an

Re: [asterisk-users] Portuguese sound files available?

Ricardo Carvalho wrote: I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Brazilian Portuguese only...

RE: [asterisk-users] Variable to show caller id for a current call?

Well, for one, you could set something like CID = ${CALLERID(number)} in the dialplan, and then GetVar CID -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: Monday, August 21, 2006 3:54 PM To: Asterisk Users Mailing List

RE: [asterisk-users] Apache for FastAGI

-Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Sunday, August 20, 2006 5:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Apache for FastAGI On Sun, Aug 20, 2006 at 04:18:11PM -0600, Douglas Garstang wrote: I'm not sure there's

[asterisk-users] Text to Speech

Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesnt cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT speech engine? The IBM ViaVoice sounds no better

Re: [asterisk-users] Text to Speech

N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. We are willing to pay for a good one that works. Anyone tried the ATT

Re: [asterisk-users] Text to Speech

Cepstral seems to sound descent...But if you have more than one voice installed (Example: different languages) I can't say it in realtime in the dialplan...I have to do a little trick like: exten = 1,1,System(/opt/swift/bin/swift -n Diane-8kHz "Hello World" -o

Re: [asterisk-users] Zaptel install - Fedora Core 5

Hi Simon, I did yum update last week and here is my current kernel: # uname -vr 2.6.17-1.2174_FC5smp #1 SMP Tue Aug 8 16:00:39 EDT 2006 # # ls -l /usr/src/kernels total 12 drwxr-xr-x 18 root root 4096 Jul 8 19:43 2.6.17-1.2145_FC5-smp-i686 lrwxrwxrwx 1 root root 26 Jul 8 19:43

[asterisk-users] Double dial dtmf sounds

Hi, I have site using only softphones (SJPhone under Windows). Once in a while the users complain that they hear double and triple dial dtmf when they dial out. What could be causing that on the asterisk side? Andre Courchesne ___ --Bandwidth

RE: [asterisk-users] Text to Speech

All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, August 21, 2006 3:23 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Voicemail and languages other than english doesn't seem to work well

I want to hear french messages. I put language=fr in the [globals] section of extensions.conf and in the [general] section of sip.conf. If I call an unavailable number, the digits are read in english even if the trace says french : -- Executing VoiceMail(SIP/103-6441, [EMAIL PROTECTED]) in

Re: [asterisk-users] Voicemail and languages other than english doesn't seem to work well

On Mon, 2006-08-21 at 23:03 +0200, Dominique Dartois wrote: I want to hear french messages. I put language=fr in the [globals] section of extensions.conf and in the [general] section of sip.conf. The french messages are at the right place : /var/lib/asterisk/sounds/fr/digits/*.gsm

Re: [asterisk-users] Text to Speech

Quoting Kevin Savoy [EMAIL PROTECTED]: Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. I like Cepstral. Using the information here:

[asterisk-users] Manager API: matching an Originate to the Newchannel event

Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track the status of that call through to completion. I can match to some

RE: [asterisk-users] Voicemail and languages other than englishdoesn't seem to work well

Thank you very much Carlos, you are absolutely right. Now it works! Thanks again. --- Dominique Dartois -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Carlos Chavez Envoyé : lundi 21 août 2006 23:09 À : Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Text to Speech

You might try runtime Dectalk for Linux available from http://www.fonix.com -- its not free, but it sounds quite nice. on Monday 08/21/2006 Time Bandit([EMAIL PROTECTED]) wrote N.B.: Please use plain text when sending to this list Can someone recommend a good text to speech engine that is

Re: [asterisk-users] Realtime and labels

Douglas Garstang wrote: Does anyone know if realtime extensions support the use of labels? I don't believe so. As I understand it, the dialplan parser internally converts n-type and labeled priorities to a straight numeric format, which is then used internally. Becuase the Realtime

Re: [asterisk-users] Text to Speech

Anytime I try and specify a voice when there is more than 1 voice in my voices directory...it has an error with the syntax you show here... Like I was saying in a previous post... - Original Message - From: Shane Young [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Re: Manager API: matching an Originate to the Newchannel event

In article [EMAIL PROTECTED], Nic Bellamy [EMAIL PROTECTED] wrote: Hi, I'm having a bit of trouble matching up Newchannel (and Newexten, etc. etc.) events with the Originate that created them. Basically, what I want to do is have software automatically initiate a call, and then track

[asterisk-users] Call file do 2 outbound call

Hello, I am not so really familar with asterisk at the moment, but i am working hard on it. Please could anybody advise me how to write a call file for the queue to do 2 outbounds call and connect both via my SIP interface. Thanks in advance. Daniel

Re: [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event

Yes there is but only in Bristuff asterisk.In bristuff when you enter Originate command you receive feedback with uniqueid of created call.So than you can trace uniqueidgreetingsmk 2006/8/21, Tony Mountifield [EMAIL PROTECTED]: In article [EMAIL PROTECTED],Nic Bellamy [EMAIL PROTECTED] wrote: Hi,

[asterisk-users] Call file do 2 outbound call

Hello, I am not so really familar with asterisk at the moment, but i am working hard on it. Please could anybody advise me how to write a call file for the queue to do 2 outbounds call and connect both via my SIP interface. Thanks in advance. Daniel

Re: [asterisk-users] Re: Manager API: matching an Originate to the Newchannel event

If you are originating a call with a Local/ channel you cannot use the uniqueID alone to track it. The only field that will follow all legs of a Local/ channel originated call is the CallerID, and that is only if you add the o flag to your Dial string. It's a very messy prospect to track calls

Re: [asterisk-users] Text to Speech

All I can find for Flite is for AAH, does it work as well with plain Asterisk? Is the setup the same? Never tried it, but it should be the same. Have a look here : http://dialogpalette.sourceforge.net/extras.html hth ___ --Bandwidth and Colocation

[asterisk-users] SLA.conf

I found this indication that Shared Line Appearance is possibly in SVN. Is it or is this just an indication that it is up and coming? http://bugs.digium.com/view.php?id=7701 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] Apache for FastAGI

First off, I don't work for Ericsson, but it is my impression that normally the call control is done in Erlang and the media processing is done ineither hardware orC.I never said you work in Ericsson ;) I You really are interested I recommend You to ask on[EMAIL PROTECTED]I did but they stopped

[asterisk-users] SIP Encryption in China

Hi all, Anyone has information on how Chinese equipment makers are encrypting the SIP signaling + media packets to avoid ISP firewalls? Recently, I was sent a sample FXS/O gateway with support for 3 flavors (Seawolf, etc) of such encryption. I don't believe they're using SIP/TLS and SRTP. At

[asterisk-users] Indonesian MFC-R2

Hi, What's wrong with T3 timed out? I use asterisk-1.2.10 package from ScopServ http://www.scopserv.com/v2/home.php?section=news (ScopServ Telephony Server 1.2.20). Here there are four pages of the scanned report from our telco http://www.flickr.com/photos/[EMAIL PROTECTED]/

[asterisk-users] Re: SIP Debug to file - Is it possible?

Hello List -I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the trick?Thanks!-- --Christopher T Aloi-- --

Re: [asterisk-users] Re: SIP Debug to file - Is it possible?

Christopher Aloi wrote: Hello List - I'm a big fan of call traces to diagnose a problem; I often use pri set debug file X to write PRI traces out to a file, anyone know of a similar method of saving IP traces (SIP,IAX) to a file? Anyone have any ngrep scripts that do the

Re: [asterisk-users] Re: SIP Debug to file - Is it possible?

Try Ethereal (I think it's called WireShark now). Does nice decoding of the packet stream to show you what's going on. Supports SIP for sure, not so sure about IAX though.-brandon On 8/21/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Christopher Aloi wrote: Hello List - I'm a big fan of call traces to

[asterisk-users] Quick, hopefully easy, question

Hey all, I've done some peeking around and can't find a GOOD listing of what the currently supported SIP headers are that Asterisk supports. My main reason is to get the CallerID/RPID settings for whether or not to display, but there's others as well. Anyone have a link? SKM

RE: [asterisk-users] Re: SIP Debug to file - Is it possible?

ngrep is also good if you only want to see SIP traffic and filter all the lower level stuff. -Original Message- From: Brandon Galbraith [mailto:[EMAIL PROTECTED] Sent: Mon 8/21/2006 8:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

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