[Asterisk-Users] automatically start application from the command prompt

Hello, Is it possible to start an asterisk application from the command prompt? This application has to dial to a number. When the calling party picks up the phone, the asterisk application had to play certain voicefiles. Kind Regards, Arjan Kroon Mobillion B.V

RE: [Asterisk-Users] automatically start application from thecommandprompt

the option Data? Kind regards, Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Collins Sent: maandag 13 februari 2006 21:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] automatically start application from

[Asterisk-Users] check call status during call

Hi, Is there a command (to use in a dial plan), to check the call status during a call. Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: [EMAIL

[asterisk-users] Play dial tone withou answer

Hi, I'm looking fore a way to play a dial tone before our IVR platform answered the phone line. I want to use for the following reason: When a caller calls our Voice Platform, the call will direct dial out to a number. I want to dial out before the inbound call is answered. But now

Re: [asterisk-users] Play dial tone withou answer

Yes Dave, We want to use to principle for the following reason. If the outbound call is not picked up, the inbound caller won't be charged for the call, because there was no answer. Arjan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent:

[asterisk-users] connect with two servers multiple time

=outboundserver context=outbound_dial_conf extensions.conf outbound_dial_conf exten = _X.,1,Dial(Zap/g1/${tel_outdial},30,${Dial_variables}) Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31

RE: [asterisk-users] Dell hardware ...

Hi, Alan, We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards in it and it works perfect. It is almost PlugPlay. greetings Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: dinsdag 12 september

[Asterisk-Users] calling out through second server.

. Thanks in advanced Arjan Kroon email: [EMAIL PROTECTED] ___ 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] Redirect to an application on other asterisk server

on asterisk00 look like: [asterisk08] type=friend username=asterisk08 secret=password context=local context=default host=dynamic Kind Regards. Arjan Kroon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] dail out with SIP through a second server

Caller asterisk00 (inbound/outbound server) SIP client (X-lite) This situation worked perfect. Thanks in advance Arjan Kroon email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] record message during dial

it is possible to use the parameters w or W. But when I place this parameter in my dail command it doesnt do anything. What am I doing wrong? Kind Regards, Arjan Kroon Mobillion B.V. email: [EMAIL PROTECTED] internet: www.mobillion.nl

[Asterisk-Users] asterisk perl commands

about perl commands Kind Regards. Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: [EMAIL PROTECTED] internet: www.mobillion.nl

[Asterisk-Users] No CDR in Macro after Dial

. But If I look in the CDR records a cant find this value. If I do an AppendCDRUserField before the dial command, the value is written in the CDR. Does anybody nows a solution for this problem? (Im using asterisk 1.0.0) Kind regards. Arjan Kroon

[Asterisk-Users] Get sysdate + 5 minutes

Hi, In my application I want to have the sysdate + 5 minutes. I know that the sysdate is in the variable ${DATTIME} But now I want to now how I get the sysdate + 5 minutes into a variable? Does anybody knows the answer? Kind Regards Arjan Kroon

[Asterisk-Users] res_perl voor asterisk 1.2.4

Can anybody tell me if this version is the right res_perl version? Kind regards. Arjan Kroon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Asterisk documentation..

I recommend the following book Asterisk The future of Telephony from O'Reilly. ISBN 0-596-00962-3 Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jack Wei Sent: dinsdag 9 mei 2006 7:56 To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] res_perl voor asterisk 1.2.4

Can anybody tell me if this version is the right res_perl version? Kind regards. Arjan Kroon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

RE: [Asterisk-Users] Generate two calls from Asterisk and bridge them

I recommended simple Meetme conference bridge http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Álvaro Palma Sent: woensdag 24 mei 2006 16:36 To: asterisk-users@lists.digium.com Subject

[Asterisk-Users] Read command

is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede email: [EMAIL PROTECTED] internet: www.mobillion.nl ___ --Bandwidth and Colocation provided

[Asterisk-Users] X-web Lite

anybody know an other embedded web based SIP client? Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: [EMAIL PROTECTED] internet: www.mobillion.nl

RE: RE : [Asterisk-Users] X-web Lite

Title: Message Hi, It is not distributed by X-ten. I found a copy on another forum. I can send it to you, if you want it. Kind regards, Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Taylor Sent: vrijdag 13 januari 2006 9:40

[Asterisk-Users] multithreading for res_perl

files it look like that asterisk will hang or freeze, if two callers calls exactly at the same time the same perl function. Does anybody now if res_perl is multithreaded? (I use res_perl 3.0 and asterisk 1.0.0) Thanx, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554

[asterisk-users] Memory leak res_perl

that there is a memory link in res_perl. Does anybody know if this is the case and maybe knows a sollution to this problem? Kind regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede email: [EMAIL PROTECTED] internet: www.mobillion.nl

RE: [asterisk-users] Memory leak res_perl

Thanks, Can you maybe give me an example of such a build-in option sebuuging. Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: maandag 10 juli 2006 11:19 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users

RE: [asterisk-users] Ping

ping Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk Sent: woensdag 6 december 2006 14:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ping Sorry to do this but I sent

RE: [asterisk-users] Calls disconnected after 1 hour

I think you have set the absolute timeout to 3600 sec. Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: donderdag 21 december 2006 6:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject

Re: [asterisk-users] Handling 3 different call ending causes

http://www.voip-info.org/tiki-index.php?page=Asterisk+variable+hangupcau se From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tobias Ahlander Sent: maandag 17 maart 2008 15:35 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users]

[asterisk-users] queue logging

Hi, I' using with asterisk a queue with tree members and round robin. When a caller enters this queue and it is connecting to one of the members, is there a possibility to see which member the caller is connected to? And is there a way to see in de application to see if the connection

Re: [asterisk-users] queue logging

Hi, I'm not looking for a programma that show the queue logging. But is there a way to check during a call, which member is connected to the caller. Kind Regard, Arjan Kroon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott

[asterisk-users] karaoke functionality

Hi, Is it possible top use a form of Karaoke Functionality? When a caller calls a number, he hears a voicefile. During this voicefile he sings along with this voicefile. Is it possible to record what the caller is singing? Grt,

Re: [asterisk-users] karaoke functionality

functionality Hi, Why not use MixMonitor(), so you have a single file with the singer and the music? Thanks. Andy On 5/20/08, Sherwood McGowan [EMAIL PROTECTED] wrote: Arjan Kroon | Mobillion wrote: Hi, Is it possible top use a form of Karaoke Functionality? When a caller calls

Re: [asterisk-users] .GSM - .WAV (or ,MP3) Conversion

Hey, I record the message in ULAW exten = s,1,Record(${A_record}:ulaw,0,60) After that I call sox with this command: /usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1 $wav_fl Regards, Arjan Kroon Mobillion BV -Oorspronkelijk bericht- Van: asterisk-users-boun

[asterisk-users] Centrale FastAgi server down

how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-users] Centrale FastAgi server down

to this machine, but fails and the application will go to the hangup clause. Is there a environment option that I can set, so that FastAgi won't go to the hangup clause, but go the the next line in the dailplan. Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun

Re: [asterisk-users] Centrale FastAgi server down

to this machine, but fails and the application will go to the hangup clause. Is there a environment option that I can set, so that FastAgi won't go to the hangup clause, but go the the next line in the dailplan. Regards, Arjan Kroon ___ -- Bandwidth

[asterisk-users] Size of Exten when using IAX

Hi, We are use IAX protocol between two asterisk servers. Now we send information through this protocol by using EXTEN We see that the variable EXTEN only holds 66 characters. If we set a value larger then 66 characters, for example 70 characters. The last 4 characters are cut off.

Re: [asterisk-users] Size of Exten when using IAX

If I look at the console (with verbosity on 3) I see that also the last 4 characters are lost. I never heard of 'wireshark on the wire' I'll try this. Is IAXVARS also supported on asterisk 1.0.0 ? -- Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

[asterisk-users] SET with pipe symbol

Hi, I want to place a pipe symbol in a variable by using the command Set I tried the following code: Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number)) When I call to my applicatie I see the following output in my CLI : Ignoring entry '612345678' with no = (and not

Re: [asterisk-users] SET with pipe symbol

Tilghman, Tx, That was the solution. Kind Regards, Arjan Kroon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: dinsdag 29 januari 2008 16:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users

[asterisk-users] Losing CALLERID{dnid}

Hi, I'm using videocalling on asterisk 1.4.10. When I setup the videocall with exten = n,1,h324m_gw([EMAIL PROTECTED]), I loose the variable DNID (${CALLERID(dnid)}) Before the videocall is set up, this variable is filled and after this videocall this variable is empty. Also all local

[asterisk-users] one CDR instead of multiple CDR

Hi, In my application I jump to different extensions For example: [begin] exten = s,1,Goto(starts,s,1) [start] exten = s,1,Play(welkom) . exten = h,1,Goto(end,s,1) [end] exten = s,1,Macro(end_call) exten = s,n, Hangup When I look at my CDR record I see three

Re: [asterisk-users] Losing CALLERID{dnid}

Sorry, I tried to use underscore(s) before the variable names, but without any success. H234m_gw is a functionality which we use for video calling on asterisk. (http://sip.fontventa.com/) -- Arjan Kroon Mobillion BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED

Re: [asterisk-users] one CDR instead of multiple CDR

and during the intro.3gp I press the #-key the call will be ended. But I got three different CDR's. Does anybody know how I can use one CDR instead of 3 different CDR's Kind Regards, Arjan Kroon Mobillion BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf

Re: [asterisk-users] Syncronizing files on different Asteriskservers

Maybe a central server is an idée. You'll have to mount an directory on server A, B and C to a directory on the central server. A disadvantage is, that you'll have to have a stable internet connection between al servers. Another solution is to make a script on the server A,B and C that

Re: [asterisk-users] Syncronizing files on different Asteriskservers

: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Syncronizing files on different Asteriskservers Thanks Alot @ Jeff LaCoursiere,@Arjan Kroon,@Robin,@Joseph @ Jeff LaCoursiere Well you already suggested that you would send all files to server A, so A is your

[asterisk-users] rawplayer in asterisk 1.0.0

; done Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] 4 PCIe cards in one asterisk server

Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan Kroon Mobillion BV

Re: [asterisk-users] 4 PCIe cards in one asterisk server

but maybe save you some worries. Christian 2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl: Hi, Does anybody have any experience with asterisk where are four PCIe cards are used in one server (TE420). So you can have max 4 * 4 * 30 channels = 480 channels used. Regards, Arjan

[asterisk-users] Dail of meetme options

Hi, I have a question about the dial command. Is the following scenario possible. 1) - Our asterisk server had a successful outbound call. - Our asterisk server has to call another caller and when answered asterisk has to connect this call to the another outbound

[asterisk-users] Use one ring-group for ISN truncs

? Example: Group 1 channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Use one group for ISN truncs

channel = 1-15,17-31 channel = 32-46,48-62 group=2 channel = 63-77,79-93 channel = 94-108,110-124 We are using the group number for the dial en originate command. For example: Zap/g3/0612345678 Regards, Arjan Kroon Mobillion BV

Re: [asterisk-users] Pbx för Windows? - Email found in subject

Mayby Freepbx. http://www.freepbx.org/ Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Christian Verzonden: 09-07-2010 14:41 Aan: asterisk-users@lists.digium.com Onderwerp: [asterisk

[asterisk-users] No CDR with originate from manager and then an redirect to a dial from manager

capability: 0x00 - SPEECH -- Called g1/0031655871460 -- DAHDI/2-1 is proceeding passing it to DAHDI/1-1 -- DAHDI/2-1 is ringing -- DAHDI/2-1 answered DAHDI/1-1 Does anybody have this same problem, or does anybody knows a solution? Asterisk Version: 1.6.2.9 Arjan Kroon Mobillion BV

Re: [asterisk-users] CDR record for call originated from CLI originate

Regards, Arjan Kroon Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens DHAVAL INDRODIYA Verzonden: 05-10-2010 09:09 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] CDR record for call originated from CLI

[asterisk-users] live audio stream in asterisk

__ast_read: Dropping incompatible voice frame on SIP/arjankroon- of format gsm since our native format has changed to 0x4 (ulaw) I'm using asterisk 1.8 Can anybody help me? Kind regards, Arjan Kroon Mobillion BV

Re: [asterisk-users] live audio stream in asterisk

(sbs) The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs The Hitz stream I don't know what kind of stream this is? Maybe someone knows this? Does anybody have an idea how the sbs stream must be streamend? Regards, Arjan Kroon Mobillion BV -Oorspronkelijk bericht- Van: asterisk

[asterisk-users] Channel name changed in asterisk 1.8

span=7,1,0,ccs,hdb3,yellow bchan=187-201,203-217 dchan=202 span=8,1,0,ccs,hdb3,yellow bchan=218-232,234-248 dchan=233 We use two seperate cards. (TE4/1/3 T4XXP (PCI)) Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation

Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'

Hi, We had the same problems. These problems accours when we try to send (from different servers) a lot of IAX calls to one server. (see couple of 100 calls at the same time) When we upgraded asterisk to version 1.8 we didn't get these problems. Regards, Arjan Kroon Van: asterisk-users-boun

[asterisk-users] Caching CALLERID(dnid)

actionid: 129675971_656137# variable: CALLERID(dnid) channel: DAHDI/11-1 Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] why does core show channels on 1.8 not show the channel

Maybe this helps: https://issues.asterisk.org/view.php?id=18603 Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jerry Geis Verzonden: 20-03-2011 21:24 Aan: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] CDR fields not being written from h extension after Dial command completes.

Hi, Does anybody have a solution to this problem? Because in this issue the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 Arjan Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.

Hi, I tried both setting (yes and no), both with the same result. Greeting, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Ishfaq Malik Verzonden: 04-04-2011 15:53 Aan: Asterisk Users Mailing

Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.

). But if I look in de Master.csv, I see that the example text is not the CDR(userfield) -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tilghman Lesher Verzonden: 05-04-2011 00:08 Aan: Asterisk Users

Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.

Hi, If I try to call out with Queue mechanism and the call is answered then hangup, the CDR(userfield) in the h exten is placed in the CDR. So for now I see that this problem only occurs with a Dial in the dialplan. -- Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun

Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.

Hi, New update. When I use the option g in a dial then the CDR fields are not updated. When I perform a dial without the option g, for example rR then the CDR field will be written in the h exten. So therefore I lose the g option in the dial. -- Arjan Kroon -Oorspronkelijk bericht

[asterisk-users] queue member invalid

= chan_alsa.so Does anybody have an idea what could be the problem? Best Regards, Arjan Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

[asterisk-users] Connected Line ID

, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Connected Line ID

We have two systems one with version 1.6 and one with version 1.8 With 1.8 we don't see the problem Unfortunately it is not possible to upgrade 1.6 to 1.8. But are there also pathes for version 1.6 Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Connected Line ID

Doug, I see that this patch is for 1.6.0.1 But we use version 1.6.2.12. And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun

Re: [asterisk-users] Connected Line ID

Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: And if I can see it, this patch is already included in version 1.6.2.12. Or am I wrong? That I can't answer. I'm still using 1.4.x and am experimenting with 1.8.x. I recall reading that it wasn't

Re: [asterisk-users] Connected Line ID

-Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 13:11 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote

Re: [asterisk-users] Connected Line ID

Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van

[asterisk-users] max one sip peer to register

. Regards, Arjan Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users

[asterisk-users] ISDN2 PCIe Card for Asterisk

Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] ISDN2 PCIe Card for Asterisk

something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon

[asterisk-users] Beep file with Record

directories: /var/lib/asterisk/sounds/beep.gsm /var/lib/asterisk/sounds/recordings/beep.gsm Regards, Arjan Kroon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] Beep file with Record

because you have to do record(foo.gsm) but you have to playback using playback(foo). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon | Mobillion Sent: Tuesday, October 04, 2011 9:21 AM To: asterisk

Re: [asterisk-users] Beep file with Record

...@lists.digium.com] Namens Andrew Latham Verzonden: 04-10-2011 16:41 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Beep file with Record On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: This is my complete CLI logging

Re: [asterisk-users] Beep file with Record

Discussion Onderwerp: Re: [asterisk-users] Beep file with Record On 10/04/2011 10:37 AM, Arjan Kroon | Mobillion wrote: exten = s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID) Hello Arjam, Did you notice that there's a missing '}' around the end of the line

Re: [asterisk-users] Beep file with Record

...@lists.digium.com] On Behalf Of Arjan Kroon | Mobillion Sent: Tuesday, October 04, 2011 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Beep file with Record Yes, In the code I use set the language exten = s,n,Set(CHANNEL(language)=nl/fvdb) So

Re: [asterisk-users] Beep file with Record

Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Beep file with Record How are you calling the beep.alaw from the dialplan? paste the relevant dialplan here and corresponding CLI logs. On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion arjan.kr

Re: [asterisk-users] Beep file with Record

and if it goes well revert back language after the recording. On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote: CLI:: -- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37, /var/lib/asterisk/sounds/recordings

Re: [asterisk-users] Beep file with Record

/sounds/recordings/serviceline/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60) exten = s,n,Set(CHANNEL(language)=nl)) On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote: Yes I already try this (only with language nl

Re: [asterisk-users] Beep file with Record

These are the directories which I gave in asterisk.conf astetcdir = /etc/asterisk astmoddir = /usr/lib64/asterisk/modules astvarlibdir = /usr/share/asterisk astdbdir = /var/spool/asterisk astkeydir = /var/lib/asterisk astdatadir = /usr/share/asterisk astagidir = /usr/share/asterisk/agi-bin

Re: [asterisk-users] Beep file with Record

Yes, That was the solution. Thanks. -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes Verzonden: 05-10-2011 10:15 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re:

[asterisk-users] Rami

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Rami

voice apps. On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote: Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

[asterisk-users] SIP jitter and packlost channel variables

. local_lostpackets = 7706 local_jitter = 2 local_maxjitter = 11 local_minjitter = 0 .. .. remote_lostpackets = 0 remote_jitter = 0 remote_maxjitter = 7 remote_minjitter = 14000 .. .. The only thing I see is this: http://www.voip-info.org/wiki/view/Asterisk+func+channel Regards, Arjan Kroon

Re: [asterisk-users] Mac OS X sip client with Video support

I'm using Bria,but X-Lite from counter path I have good result with these programs under Lion On 26 Apr 2012, at 12:05 PM, Alex Balashov wrote: Have you looked into Blink? On 04/26/2012 05:41 AM, Paolo Supino wrote: Hi I'm looking for a SIP client for Mac OS X (I'm running Lion) that

[asterisk-users] asterisk crashed on segmentation fault

the following drives and asterisk: Asterisk 1.6.2.12 libpri 1.4.11.4-1_centos5 dahdi linux-2.4.0-1_centos5 We are using two Digium, Inc. Wildcard TE420P quad-span T1/E1/J1 card 3.3V (PCI-Express) (rev 02) Kind regards, Arjan Kroon

[asterisk-users] Channelvariables not set in Newchannel event and DADHIChannel

event. This seems like a bug to me, or is there a purpose for this behaviour? Anyway, receiving channelvariables in Newchannel- and DAHDIChannel- events would be a very useful functionality for my application. Best regards, Arjan Kroon Below you'll find some information about my testcase