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

/0031650747314-3): CHANNEL(uniqueid)=dijkivr04-nwg.mobillion.biz-1353083131.2 ChanVariable(DAHDI/i1/0031650747314-3): AJ_TRACE_ID=AJ_ORIGINATE_2 -- above: AJ_TRACE_ID set Regards, Arjan Kroon Mobillion BV -- _ -- Bandwidth and Colocation

[asterisk-users] asterisk crashed on segmentation fault

Hello, I have a problem. One every couple of months my asterisk system crashes with a segmentation fault. kernel: asterisk[20527]: segfault at 0808 rip 2aaac952d8f2 rsp 40edb910 error 4 (This is in /var/log/messages) If I look at the same timestamp in the warning log

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] 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

[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

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] Beep file with Record

Hi, I'm using the functionality Record in asterisk 1.8.5. But when I want to record something I get the following error message: file.c:644 ast_openstream_full: File beep does not exist in any format Could anybody tell me where I have to place the beep.gsm file? I already tried the following

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

[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] max one sip peer to register

Hi, Is there a easy way to configure the sip settings so it is not possible to register more than one sip user with the same username/password. Now it is possible to register more than one sip user with the same username/password. So if I call that sip user, both sip clients will ring.

Re: [asterisk-users] Connected Line ID

...@lists.digium.com] Namens Doug Lytle Verzonden: 10-06-2011 14:01 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: But are there also pathes for version 1.6 The last patch available for the 1.6 series

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] 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

[mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 10-06-2011 12:58 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Connected Line ID Arjan Kroon | Mobillion wrote: Does anybody have problems with a wrong Connected Line ID

[asterisk-users] queue member invalid

Hi, I'm using asterisk version 1.8.3.3. In earlier versions I used queues, but with the new version the queuing mechanism doesn't work If I look in the CLI at I see that the queue-member is invalid: Members: DADHI/g3/0655871460 (Invalid) has taken no calls yet The

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

Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue

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

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion Verzonden: 05-04-2011 08:21 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. Hi

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

- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion Verzonden: 05-04-2011 09:32 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after

[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

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

List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes. On Mon, 2011-04-04 at 13:58 +0200, Arjan Kroon | Mobillion wrote: Hi, Does anybody have a solution to this problem? Because in this issue

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] 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] 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] 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] 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] 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] CDR record for call originated from CLI originate

Hi Dhaval, I 'm in the almost same situation. I've already post a issue with asterisk. https://issues.asterisk.org/view.php?id=17826 Is you only use an originate and not an originate en then redial maybe this link helps you further. https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes

[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] 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-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

[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] 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] 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

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

I don't know if you server is running under Unix. If so, here is a wiki link about mounting http://en.wikipedia.org/wiki/Mount_%28Unix%29 Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens ABBAS SHAKEEL Verzonden: 21-10-2009 08:59 Aan:

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

[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

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Michiel van Baak Verzonden: 26-06-2009 11:29 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [asterisk-users] Centrale FastAgi server down On 10:42, Fri 26 Jun 09, Arjan Kroon | Mobillion wrote: Hi, How do you all handle

Re: [asterisk-users] Centrale FastAgi server down

Hi, Your correct, this the best way. But we don't have any 'balancing' on the localhost. In some cases we have to connect directly to a central database. (we have only one central database) If the machine where the central database is running on, is down, than FastAgi will try to connect to

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] 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] queue logging

Wolfe Sent: woensdag 9 april 2008 17:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue logging You could ASTassistant to see this. Its Freeware. www.astassistant.com - Original Message - From: Arjan Kroon | Mobillion

[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] 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]

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

[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] SET with pipe symbol

] SET with pipe symbol On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote: 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

[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] Size of Exten when using IAX

] On Behalf Of Tilghman Lesher Sent: dinsdag 30 oktober 2007 15:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Size of Exten when using IAX On Tuesday 30 October 2007 08:40:51 Arjan Kroon | Mobillion wrote: We are use IAX protocol between two asterisk

[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.