[asterisk-users] Channelvariables not set in Newchannel event and DADHIChannel
I'd like to receive channelvariables in my AMI-events. Unfortunately this seems not to work for Newchannel- and DAHDIChannel- events caused by an Originate. In manager.conf I putted the following line: channelvars=CHANNEL(uniqueid),AJ_TRACE_ID . Only CHANNEL(uniqueid) appears to work for every 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. Asterisk 1.8.15 settings in manager.conf [general] channelvars=CHANNEL(uniqueid),AJ_TRACE_ID flow 1) originate with __AJ_TRACE_ID=AJ_ORIGINATE_2 2) Newchannel event received without AJ_TRACE_ID set 3) DAHDIChannel event received without AJ_TRACE_ID set 4) NewAccountCode event received and AJ_TRACE_ID is set action: Originate actionid: 608078545_40#AJ_ORIGINATE_2 callerid: async: true variable: CALLERPRES()=prohib variable: dcoModURL= variable: dcoCoreURL=http://localhost:8080/DijkConnectIVRWebApp/services/TestService variable: dcoLicenseeId=39 variable: dcoApplicationReference= variable: dcoPresentationNumber= variable: dcoCallId=TESTMOBILLION3 variable: dcoRecordingStorageReference= variable: dcoRedirectCallId= variable: dcoSetupTimedOut=SETUP variable: dcoAgentId=100 variable: dcoDestinationNumber=0031650747314 variable: __AJ_TRACE_ID=AJ_ORIGINATE_2 variable: dcoRecordingMode= variable: dcoConnectionTimeOut=60 variable: dcoUserRole=OUTBOUND_AGENT variable: dcoRecordingProcessingURL= variable: dcoDialMode=AGENT variable: dcoRingTimeOut=60 variable: dcoRecordingStorageDuration= variable: dcoBillingReference=t=mobtest priority: 1 exten: s context: setup_agent channel: DAHDI/g1/0031650747314 timeout: 13 message : Originate successfully queued response : Success actionid : 608078545_40#AJ_ORIGINATE_2 Event: Newchannel Privilege: call,all SequenceNumber: 1877 File: channel.c Line: 1349 Func: __ast_channel_alloc_ap Channel: DAHDI/i1/0031650747314-3 ChannelState: 1 ChannelStateDesc: Rsrvd CallerIDNum: CallerIDName: AccountCode: Exten: Context: incoming Uniqueid: dijkivr04-nwg.mobillion.biz-1353083131.2 ChanVariable(DAHDI/i1/0031650747314-3): CHANNEL(uniqueid)=dijkivr04-nwg.mobillion.biz-1353083131.2 ChanVariable(DAHDI/i1/0031650747314-3): AJ_TRACE_ID= -- above: AJ_TRACE_ID not set Event: DAHDIChannel Privilege: call,all SequenceNumber: 1878 File: chan_dahdi.c Line: 2141 Func: dahdi_ami_channel_event Channel: DAHDI/i1/0031650747314-3 Uniqueid: dijkivr04-nwg.mobillion.biz-1353083131.2 DAHDISpan: 1 DAHDIChannel: 1 ChanVariable(DAHDI/i1/0031650747314-3): CHANNEL(uniqueid)=dijkivr04-nwg.mobillion.biz-1353083131.2 ChanVariable(DAHDI/i1/0031650747314-3): AJ_TRACE_ID= -- above: AJ_TRACE_ID not set action: GetVar actionid: 608078545_41# variable: AJ_TRACE_ID channel: DAHDI/i1/0031650747314-3 response : Success actionid : 608078545_41# value : AJ_ORIGINATE_2 variable : AJ_TRACE_ID Event: NewAccountCode Privilege: call,all SequenceNumber: 1900 File: cdr.c Line: 1010 Func: ast_cdr_setaccount Channel: DAHDI/i1/0031650747314-3 Uniqueid: dijkivr04-nwg.mobillion.biz-1353083131.2 AccountCode: OldAccountCode: ChanVariable(DAHDI/i1/0031650747314-3): 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 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 file of asterisk (/var/log/asterisk/warning), I see that the are warning about fix up channel: WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/2 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/6 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 6 to 8 because 8 is already in use WARNING[24000] chan_dahdi.c: Ringing requested on channel 0/8 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 3 to 4 because 4 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/4 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 2 to 3 because 3 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/3 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 4 to 5 because 5 is already in use WARNING[24000] chan_dahdi.c: Hangup REQ on bad channel 0/5 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Answer requested on channel 0/2 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Answer requested on channel 0/6 not in use on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 1 to 2 because 2 is already in use WARNING[24000] chan_dahdi.c: Hangup on bad channel 0/2 on span 1 WARNING[24000] chan_dahdi.c: Can't fix up channel from 2 to 3 because 3 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 3 to 4 because 4 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 4 to 5 because 5 is already in use WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use WARNING[24000] chan_dahdi.c: Hangup on bad channel 0/6 on span 1 WARNING[24000] chan_dahdi.c: Whoa, there's no owner, and we're having to fix up channel 6 to channel 8 WARNING[24000] chan_dahdi.c: Whoa, there's no owner, and we're having to fix up channel 1 to channel 2 WARNING[24000] chan_dahdi.c: Can't fix up channel from 5 to 6 because 6 is already in use [/size] Does anybody know what these messages mean? I use 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 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 has video support. I've tried Linphone but for the life of me I can't get it to add a sip account (the apply button is always grayed out) :-( Can anyone recommend other SIP clients that have video Support for Mac OS X? TIA Paolo -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP jitter and packlost channel variables
Hi,
A client of ours get lots of problem with there voice quality when the do a lot
SIP calls.
In a application I log the rtpqos audio jitter an lost packets. (see Below)
Does anybody know what the numbers mean?
If I look at a sample of the channel variables, I see the following number.
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
Mobillion BV
exten = s,n,Set(A_SIP_DATA=${CHANNEL(rtpqos,audio,local_lostpackets)})
exten = s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_jitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_maxjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_minjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_normdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_stdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_lostpackets)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_jitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_maxjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_minjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_normdevjitter)})
exten =
s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_stdevjitter)})
--
_
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[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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rami
Is this freeware, or a module which you can include in your ruby code? Or is it a complete framework? On 04 Jan 2012, at 5:31 PM, gokulnath wrote: Hey, There is a new kid in town if you want to code in ruby. Use adhearsionhttps://github.com/adhearsion/adhearsion/wiki, it's a framework to make 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards Gokulnath @8129845320 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-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 9:43 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 therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
line/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
It is
(IMO) somewhat confusing 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-users@lists.digium.com
Subject: [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 directories:
/var/lib/asterisk/sounds/beep.gsm
/var/lib/asterisk/sounds/recordings/beep.gsm
Regards,
Arjan Kroon
Beep is called from
http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it looks fine
a first glance. Are you using the language prefix?
--
~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
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--
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Re: [asterisk-users] Beep file with Record
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:04
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...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@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 therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@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.nlmailto:arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/service
line/${UNIQUEID) exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep).
It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but
you have to playback using playback(foo).
-Original Message-
From:
asterisk-users-boun
Re: [asterisk-users] Beep file with Record
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to en 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/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:04
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...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx should just be 2 characters,
not xx/ (nl/fvdb) (feel free to correct my assumption that language has not
been expanded beyond the 2 character limitation)?
-Original Message-
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@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 therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@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.nlmailto:arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,6
0) in new stack [Oct 4 16:19:38] WARNING[13370]: file.c:644
ast_openstream_full: File beep does not exist in any format [Oct 4
16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory [Oct 4 16:19:38]
WARNING[13370]: app_record.c:281 record_exec: ast_streamfile failed on
CAPI/ISDN1#02/318647615-37
In de Conf file I use the following
Re: [asterisk-users] Beep file with Record
Oke, I tried this, but sorry
-- Executing [s@servicelijn:91] Set(CAPI/ISDN1#02/318647615-3e,
CHANNEL(language)=en) in new stack
-- Executing [s@servicelijn:92] Set(CAPI/ISDN1#02/318647615-3e,
A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/1317800460.74)
in new stack
-- Executing [s@servicelijn:93] Record(CAPI/ISDN1#02/318647615-3e,
/var/lib/asterisk/sounds/recordings/serviceline/1317800460.74.wav,0,60) in
new stack
[Oct 5 09:41:03] WARNING[18963]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 5 09:41:03] WARNING[18963]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 5 09:41:03] WARNING[18963]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-3e
== Spawn extension (servicelijn, s, 93) exited non-zero on
'CAPI/ISDN1#02/318647615-3e'
This is my conf.file
exten = s,n,Set(CHANNEL(language)=en)
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID})
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Sammy Govind
Verzonden: 05-10-2011 09:32
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
hmmm...what i'm saying is this
exten = s,n,Set(CHANNEL(language)=en))
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
exten = s,n,Set(CHANNEL(language)=nl))
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:26
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
Since you've changed the language (sound directory) So just as a test change
the language back to en and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
I don't call the beep file in my dialplan.
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Sammy Govind
Verzonden: 05-10-2011 09:04
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...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same result.
-Oorspronkelijk bericht-
Van:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
Namens Danny Nicholas
Verzonden: 04-10-2011 17:16
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
I see two problems here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4 thing?) Set(CHANNEL(language)=xx) xx
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 astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk I try to change de astdatadir into /var/lib/asterisk/ But when I restart asterisk and I look at the settings in the CLI I still see Data directory: /usr/share/asterisk How can I reload the new settings? -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes Verzonden: 05-10-2011 09:50 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Beep file with Record Hi Arjan, I also try to place the voicefile in the directory /var/lib/asterisk/sounds/ and /var/lib/asterisk/sounds/applications/ of but without any success. Just for double-checking, but what directory is listed as the astdatadir in asterisk.conf? Best regards, Jeroen Eeuwes -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Arjan, I try to change de astdatadir into /var/lib/asterisk/ But when I restart asterisk and I look at the settings in the CLI I still see Data directory: /usr/share/asterisk At least that explains why it can't find your beep-file. It is looking in /usr/share/asterisk and not /var/lib/asterisk. If your asterisk.conf says this: [directories](!) ; remove the (!) to enable this you should remove the (!) to enable the alternate directories in asterisk.conf so it should only say this: [directories] Best regards, Jeroen Eeuwes -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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 us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beep file with Record
This is my complete CLI logging
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: ast_streamfile
failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep). It is
(IMO) somewhat confusing 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-users@lists.digium.com
Subject: [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 directories:
/var/lib/asterisk/sounds/beep.gsm
/var/lib/asterisk/sounds/recordings/beep.gsm
Regards,
Arjan Kroon
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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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 therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open
beep (format 0x8 (alaw)): No such file or directory
[Oct 4 16:19:38] WARNING[13370]: app_record.c:281 record_exec:
ast_streamfile failed on CAPI/ISDN1#02/318647615-37
In de Conf file I use the following command:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk-users] Beep file with Record
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep). It is
(IMO) somewhat confusing 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-users@lists.digium.com
Subject: [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 directories:
/var/lib/asterisk/sounds/beep.gsm
/var/lib/asterisk/sounds/recordings/beep.gsm
Regards,
Arjan Kroon
Beep is called from
http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it
looks fine a first glance. Are you using the language prefix?
--
~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~
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Re: [asterisk-users] Beep file with Record
Yes,
Copy past error in mail.
In the code it is correct.
sorry
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jose P. Espinal
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial 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 (on
the UNIQUEID part)?
--
# Jose P. Espinal
# http://www.eSlackware.com
# IRC: Khratos @ #asterisk / -doc / -bugs
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[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://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN2 PCIe Card for Asterisk
Hi Tamar, Yes, I mean 1 Port ISDN BRI PCIe board. We need an PCIe board, because the board don't provide PCI slots, only PCIe slots. It doesn't matter which distribution we use. But I will look at Sangoma. Best Regards, Arjan Mobillion BV -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Tamer Higazi Verzonden: 06-09-2011 10:39 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] ISDN2 PCIe Card for Asterisk what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- € inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need 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 Mobillion. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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...@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 was for 1.6.0.1: https://issues.asterisk.org/jira/browse/8824 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connected Line ID
Oke, But is there a patch from version 1.6.2.12? Greeting, Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial 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 supported directly until 1.8 without patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connected Line ID
succeeded at 1846 with fuzz 1 (offset 195 lines). 2 out of 3 hunks FAILED -- saving rejects to file include/asterisk/channel.h.rej patching file include/asterisk/frame.h Hunk #1 FAILED at 84. Hunk #2 FAILED at 300. 2 out of 2 hunks FAILED -- saving rejects to file include/asterisk/frame.h.rej patching file main/channel.c Hunk #1 succeeded at 1444 (offset 190 lines). Hunk #2 succeeded at 1338 (offset 5 lines). Hunk #3 FAILED at 2867. Hunk #4 FAILED at 3357. Hunk #5 FAILED at 3398. Hunk #6 FAILED at 4298. Hunk #7 succeeded at 6159 with fuzz 2 (offset 963 lines). 4 out of 7 hunks FAILED -- saving rejects to file main/channel.c.rej patching file main/dial.c Hunk #1 succeeded at 274 with fuzz 1 (offset 1 line). Hunk #3 succeeded at 430 (offset 1 line). patching file main/features.c Hunk #1 succeeded at 4566 with fuzz 2 (offset 1271 lines). patching file main/rtp.c Hunk #1 FAILED at 3389. Hunk #2 FAILED at 3630. 2 out of 2 hunks FAILED -- saving rejects to file main/rtp.c.rej -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: But is there a patch from version 1.6.2.12? Not that I can see. You could try applying the patches against that version and see if they apply cleanly. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connected Line ID
Ryan, The problem is not with SIP, but with ISDN. Or is this patch also applied for ISDN calls? Arjan -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Ryan Wagoner Verzonden: 20-06-2011 13:51 Aan: Asterisk 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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle Verzonden: 20-06-2011 11:36 Aan: Asterisk Users Mailing List - Non-Commercial 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 supported directly until 1.8 without patches. Doug I am using 1.8 now, but I had updated the patch for SIPCalledRPID() for 1.6.2 and was using it successfully. http://pastebin.com/K1mmGU1c Ryan -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connected Line ID
Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the 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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 [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 with asterisk version 1.6 As far as I know, unless you're applying patches yourself, Connected Line ID is only available for the 1.8 series. I'm running it on 1.4 with patches. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 queues.conf looks like this: [general] persistentmembers = yes monitor-type = MixMonitor [test] musicclass = default strategy = rrmemory member = DADHI/g3/0655871460 timeout = 60 retry = 1 maxlen = 5 If already changed the modules.conf to this, but with no success [modules] autoload=yes preload = pbx_config.so preload = pbx_ael.so preload = chan_local.so preload = app_queue.so noload = pbx_gtkconsole.so load = res_musiconhold.so noload = 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 Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR fields not being written from h extension after Dial command completes.
Hi, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). 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 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 the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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...@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, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). 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 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 the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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- 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 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...@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, I don't use a macro. I stay in the same dialplan (application) In the h exten I place a test (for example testThis is a test/test) If I look at the CLI and after I placed the example text in the variable CDR(Userfield), I see (with NoOp) that example text is placed in CDR(Userfield). 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 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 the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 The h extension should be in the context from which the Macro was called, not in the Macro context itself. -- Tilghman -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 the solution is not mentioned. https://issues.asterisk.org/view.php?id=18522 Arjan Kroon Hi Have you set endbeforehexten=yes in your cdr.conf? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Discussion Onderwerp: [asterisk-users] why does core show channels on 1.8 not show the channel When I do core show channels on 1.8 it gives me something like: Channel Location State Application(Data) DAHDI/i1/3175551212- s@default:10 Up BackGround(SM_ATTENDANT) 1 active channel 1 active call 188 calls processed No active MeetMe conferences. What channel is i1?? It used to show me DAHDI/18/3175551212 . How do I relate i1 to 18 which is the real channel. Jerry -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caching CALLERID(dnid)
Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read the variable CALLERID(dnid) with AMI. This call will be ended. - Then we make an outbound call on the same channel. The CALLERID(dnid) is not set, during this outbound call. If this outbound call is picked up, we will read the CALLERID(dnid) with AMI. Now we see that the CALLERID(dnid) is still '655871460' Is there a way to reset the CALLERID(dnid) on one channel or automatically reset the complete cache on one channel if this channel is ended? Regards, Ami command: action: GetVar 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 every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonas Kellens Verzonden: 14-01-2011 14:31 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' On 01/14/2011 02:22 PM, Thorsten Göllner wrote: Am 14.01.2011 12:50, schrieb Jonas Kellens: On 01/14/2011 12:44 PM, Thorsten Göllner wrote: Am 14.01.2011 11:55, schrieb Jonas Kellens: Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' snip [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:19] DEBUG[27654] channel.c: Failure, could not lock '0x114af2c0' after 199 retries! Question 1 : What can be causing this ?? Question 2 : What can I do when this happens ? Because Asterisk was no longer responding untill I rebooted the server. What is the right way to handle this extreem situation ? Question 3 : How can I avoid this situation from happening again ? Sometimes I can see this messages too - but with no impact. It is a debug-message and should not indicate any problems. What does it mean when you say Asterisk was no longer responding? -Thorsten- Hello, the debug-file is flooded with this message during 2 à 3 seconds and counts about 300 à 400 lines... So I don't think it's just a debug-message. Asterisk was not responding as in core show channels had no output, sip show peers had no output, core restart now did nothing... The Asterisk proces was still running though... Also: all registrations of SIP peers were lost. I could see that the IP-phones lost their registration to the Asterisk server. And they did not re-register untill the server was finally rebooted. This message is repeated over 100 times. (You can take a look at the source code.) Which Asterisk-Version do you use? Did it happen before or again? -Thorsten- Hello, I use asterisk 1.6.2.10 As I said, this is the first time I experience this. I used 1.4 before, never had this. I'm using 1.6.2.10 now for about 5 à 6 months and this is the first time. Kind regards, Jonas. -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel name changed in asterisk 1.8
Hi, The channel name for DAHDI channels has changed in 1.8 with no information that I can find in the ChangeLog. The old format was DAHDI/XX-Y where XX was the real channel number. It has changed to DAHDI/iZ/XX-YYY where XX is the callerid. And Z is the number of the span in /etc/dahdi/system.conf Our channel names look like this DAHDI/i8/0517383600-229 DAHDI/i1/0031650545840-329 DAHDI/i4/0512515245-20f DAHDI/i6/0517417488-1fb But we want to know which channel number of these four channels is used. P.S. This is our system.conf: span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,yellow bchan=32-46,48-62 dchan=47 span=3,1,0,ccs,hdb3,yellow bchan=63-77,79-93 dchan=78 span=4,1,0,ccs,hdb3,yellow bchan=94-108,110-124 dchan=109 span=5,1,0,ccs,hdb3,yellow bchan=125-139,141-155 dchan=140 span=6,1,0,ccs,hdb3,yellow bchan=156-170,172-186 dchan=171 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 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] live audio stream in asterisk
Hi Daniel/asterisk users, You're correct, a typo. If got now to stream configured in musiconhold.conf [Hitz] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-dtc-aa02.stream.aol.com:80/stream/1074 [sbs] mode=custom application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx If I try to play the Hitz stream, it works correctly and if I try to play the sbs stream I hear nothing? exten = s,n,MusicOnHold(Hitz) or exten = s,n,MusicOnHold(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-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Daniel Tryba Verzonden: 24-12-2010 16:12 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] live audio stream in asterisk On Fri, Dec 24, 2010 at 02:36:40PM +0100, Arjan Kroon | Mobillion wrote: Is it possible to use a live audio stream in asterisk Yes, there are examples on: http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf#Exampleusingasxmmswmvstreamsoranythingth BTW You have a typo in your config (custum should be custom). -- Daniel Tryba -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] live audio stream in asterisk
Hi, Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I did not get it working. This is my musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [general] [default] mode=custum directory=/var/lib/asterisk/mohmp3/stream,http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx This is my extension.conf exten = _X.,1,Answer exten = _X.,n,MusicOnHold() If I look in the CLI I get the following error: Executing [...@test_moh:2] MusicOnHold(SIP/arjankroon-, ) in new stack -- Music class default requested but no musiconhold loaded. [Dec 24 14:34:03] NOTICE[9030]: channel.c:4006 __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 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 originate hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No CDR with originate from manager and then an redirect to a dial from manager
Hi, The ami manager call out with an originate through dadhi to a local number (A). If this call is answered, then the ami manager redirect this call to a dial command. This dial command calls through dadhi to another local number (B). Number B answers this call and number A en B are connected. If number B and number A hangs up, there is will be no CDR be written If the dial command is commented out, (so there is no dial to number B), a CDR will be written. I think this bug is referring to issue https://issues.asterisk.org/view.php?id=17592nbn=16 [^https://issues.asterisk.org/view.php?id=17592nbn=16] The path in this issue is installed on our servers. Additional Information: Logging of the Dial command which was used to call number B. (after the redirect) Executing [...@setup_agent:229] Dial(DAHDI/1-1, DAHDI/g1/0031655871460,30,tgR) in new stack -- Requested transfer 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 -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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] Pbx för Windows? - Email found in subject Hi all, Yes, this is not the right list for such a question and I am using Asterisk myself its for a friend who isn't used to Linux. You can write me off list if you want. He is looking for a Windows based PBX with same functionality as Asterisk. Any tips? Many thanks! -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use one ring-group for ISN truncs
Hi, A question. We are using TE420 cards. Normally we configure for each truncs one ring-group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more ring-groups to one ring-group? 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 to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use one group for ISN truncs
Hi, A question. We are using TE420 cards. Normally we configure for each truncs one group. group=1 channel = 1-15,17-31 group=2 channel = 32-46,48-62 group=3 channel = 63-77,79-93 group=4 channel = 94-108,110-124 My question now, is it possible to join more groups to one group? 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 We are using the group number for the dial en originate command. For example: Zap/g3/0612345678 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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 call. My first question is , do I have to this with a DIAL command, of a MEETME command? (A) - When both party a connected it must be possible to disconnect the other party, but the line must not be hanged up. (it must be possible to play a sound file to this 'disconnected' party? (B) 2) - Our asterisk server had a successful outbound call. - A new caller is calling our asterisk server and when answered asterisk has to connect this call to the another outbound call. My first question is , do I have to this with a DIAL command, of a MEETME command? (C) - When both party a connected it must be possible to disconnect the other party, but the line must not be hanged up. (it must be possible to play a sound file to this 'disconnected' party? (D) Could anybody help me if these scenario's works? Best Regards, A.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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 4 PCIe cards in one asterisk server
Hi, We are now using 2 PCI cards (TE410) in all our server without any problem. Because we want to reduce the power consumention of the complete server-park, we though to put 4 PCIe cards in 1 server. We have a redundancy of our servers, so machine fails is not a great issue. Regards, Arjan Kroon -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Christian Victor Verzonden: 22-02-2010 15:22 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] 4 PCIe cards in one asterisk server Not wit four - but two of them in a single core 3GHz machine worked flawlessly doing only switching and IVR without codec conversion. Many will suggest that you split your lines on two machines to to prevent a total loss when a machine fails. This will add some work on setup 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 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/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rawplayer in asterisk 1.0.0
Hi, We are using asterisk version 1.0.0. For queue'ing we use the rawplayer script to play a music file in the background. Now we see that after a while all the sessions on our Linux environment will be taken by the rawplayer process. An example of such a session is (done with ps -ax|grep rawplayer) 24785 ?Z 0:00 [rawplayer defunct] 8415 ?Z 0:00 [rawplayer defunct] 13821 ?Z 0:00 [rawplayer defunct] 18868 ?Z 0:00 [rawplayer defunct] 22950 ?Z 0:00 [rawplayer defunct] The only thing to get rid of these sessions is to restart asterisk and then kill all rawplayer sessions Does anybody have the same problem with this problem. A way is to upgrade asterisk, but this is not now the solution for us. The code for the rawplayer is: /usr/bin/rawplayer #!/bin/sh for name in $@; do cat $name ; 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/mailman/listinfo/asterisk-users
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: 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 server Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . Can you give me some clues or links @ Arjan Kroon As i dont have good idea about Mounting what about the script actually i need some thing that dont needs human hand after development. And if script can do this then it will be fine. @Robin Which Application do use for that ?? Please elaborate Hell, you could even abuse dropbox for this purpose. What does this means? @ Joseph No Joseph its not some thing voice mail its recording of suggestions etc Actually operators are located at different locations and if a user leave a suggestion at one operator then the file will be on that particular server. But if the user of another operator want to listen that file then this file must be present on that server also ..Thats why I am considering these options On Wed, Oct 21, 2009 at 10:08 AM, Joseph syscon...@gmail.com wrote: On 10/20/09 17:24, ABBAS SHAKEEL wrote: Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C users come to listen the recorded voices (I am using agi ). any user records his profile on server B , NOW a user who make a call to server C cannot listen to profiles recorded at server B. Because these profiles reside on Server B ... Similar in case of server C. By ...listen to profile... do you mean retrieve their voice-mail on a different server? -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 copies the recorder files to the others servers. Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens ABBAS SHAKEEL Verzonden: 20-10-2009 15:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Syncronizing files on different Asteriskservers Yeah i do have considered that option but the problem is that in if i have four servers server ie A,B,C,D... all cannot be servers ands clients at the same time. Thats the reason I am wondering any other solution On Tue, Oct 20, 2009 at 5:44 PM, Jeff LaCoursiere j...@jeff.net wrote: On Tue, 20 Oct 2009, ABBAS SHAKEEL wrote: Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C users come to listen the recorded voices (I am using agi ). any user records his profile on server B , NOW a user who make a call to server C cannot listen to profiles recorded at server B. Because these profiles reside on Server B ... Similar in case of server C. I thought a solution that i will use sockets. when a user records a voice on Server B . The file will be send to Server A and Server A will send it to all other servers ie C and others if exists. But if alot of user start to record their voices then sockets may fail ??? DO any one have idea to do it in better way ??? How about rsync? j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Centrale FastAgi server down
Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints Unable to locate host and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about 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/mailman/listinfo/asterisk-users
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 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...@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 the situation when a centrale fastagi server process(es) are down? AGI(..) prints Unable to locate host and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? Let it connect to localhost and use balance to handle the connection to a set of fastcgi servers so you have redundancy :) 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/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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
$wav_fl
Regards,
Arjan Kroon
Mobillion BV
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Tim Dobson
Verzonden: 14-04-2009 13:39
Aan: asterisk-users@lists.digium.com
Onderwerp: [asterisk-users] .GSM - .WAV (or ,MP3) Conversion
Hey there,
I'm trying to convert some call recordings from asterisk we have in .gsm
format to something I can pipe through ffmpeg - wav would be good, mp3
would be amazing!
I've been trying playing with sox but I don't seem to be getting too far
with
1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
as ffmpeg borks at it:
t...@freee-meee:~/dmc/call recordings$ ffmpeg -i 1239101491.30.conv.wav
1239101491.30.conv.mp3
FFmpeg version r11872+debian_3:0.svn20080206-12ubuntu3.1, Copyright (c)
2000-2008 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-pp --enable-swscaler
--enable-x11grab --prefix=/usr --enable-libgsm --enable-libtheora
--enable-libvorbis --enable-pthreads --disable-strip --enable-libfaad
--enable-libfaadbin --enable-liba52 --enable-liba52bin
--enable-libdc1394 --disable-armv5te --disable-armv6 --disable-altivec
--disable-vis --enable-shared --disable-static
libavutil version: 49.6.0
libavcodec version: 51.50.0
libavformat version: 52.7.0
libavdevice version: 52.0.0
built on Mar 13 2009 17:48:10, gcc: 4.3.2
Input #0, wav, from '1239101491.30.conv.wav':
Duration: 00:00:06.7, bitrate: 1040 kb/s
Stream #0.0: Audio: libgsm_ms, 64 Hz, mono, 1040 kb/s
File '1239101491.30.conv.mp3' already exists. Overwrite ? [y/N] y
Output #0, mp2, to '1239101491.30.conv.mp3':
Stream #0.0: Audio: mp2, 64 Hz, mono, 64 kb/s
Stream mapping:
Stream #0.0 - #0.0
[mp2 @ 0xb7d352f0]Sampling rate 64 is not allowed in mp2
Error while opening codec for output stream #0.0 - maybe incorrect
parameters such as bit_rate, rate, width or height
t...@freee-meee:~/dmc/call recordings$
Has anyone got any suggestions based on previous experience?
www.tdobson.net
If each of us have one object, and we exchange them, then each of us
still has one object.
If each of us have one idea, and we exchange them, then each of us now
has two ideas. - George Bernard Shaw
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[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, image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] karaoke functionality
Yes, Thanks, Monitor() was the solution. It works perfect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Kuo Sent: woensdag 21 mei 2008 5:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] karaoke 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 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, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, this is entirely possible using Monitor(). http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor When you record a conversation using the monitor command, the end result is two files, a [name]-in.[ext] file and a [name]-out.[ext] fileI believe you're looking for the input side, I always get them confused Just be sure not to use the m option, that would mix the two channels together into a single sound file. Hope this helps, Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com Sent: Wednesday, April 09, 2008 1:01 AM Subject: [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 from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 from the caller to one of the members was successful of not successful? I know you can see it in de queue. log. But I want to know if I can see it also in the hangup (h) in de application? Kind Regards image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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] Handling 3 different call ending causes
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply
the ${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the
instructions to continue executing to the next priority for that
extension, whereas if the call completes (Dial() is successfully
connected), this does not happen.
I''ve tried to use the h extension in combination with the ${CHANNEL} in
the dialplan as suggested on the wiki page, but I haven't had any luck
with it.
For this test I have a Sipura phone with number 1003 and a X-lite with
1203. If I let the time go by (the 20 seconds defined in the Dial
Command) I get the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup
is SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup
is SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup
is SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small context
with the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
Thanks again,
Best regards,
Tobias
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Re: [asterisk-users] Losing CALLERID{dnid}
Sorry,
I tried to use underscore(s) before the variable names, but without any
success.
H234m_gw is a functionality which we use for video calling on asterisk.
(http://sip.fontventa.com/)
--
Arjan Kroon
Mobillion BV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: dinsdag 5 februari 2008 1:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Losing CALLERID{dnid}
On Mon, 2008-02-04 at 10:08 +0100, Arjan Kroon | Mobillion wrote:
When I setup the videocall with exten =
n,1,h324m_gw([EMAIL PROTECTED]), I loose the variable DNID
(${CALLERID(dnid)})
Hmmmn... I'm not familiar with the h324m_gw application. Is that some
third-party add-on to Asterisk?
Have you tried doing something like:
exten = blah,1,Set(__MY_DNID=${CALLERID(dnid)})
exten = blah,n,h324m_gw([EMAIL PROTECTED])
and see if that MY_DNID channel variable is still set after the call?
(The underscores on the beginning of the variable tell Asterisk that any
child channels should inherit the channel variable from this channel.)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] one CDR instead of multiple CDR
This is a part of our programma. [begin] exten = s,1, h324m_gw([EMAIL PROTECTED]) [video] exten = s,1,h324m_gw_answer() exten = s,2,Wait(3) exten = s,3,Goto(intro,s,1) [intro] exten = s,1,mp4play(intro.3gp) exten = #,n,Goto(einde,s,1) [einde] exten = s,n, Hangup() When I use this dialplan 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 Of Atis Lezdins Sent: maandag 4 februari 2008 15:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] one CDR instead of multiple CDR On 2/4/08, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote: 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 different CDR's in my record. Is there a way to use one CDR on every call, and not multiple CDR on every call? You should post also the relevant sections of your dialplan that manipulates CDR's. For example calls to Dial() or Queue() applications. Also a log snippets (uncomment the full line in logger.conf) that says anything about posting CDR and previous few commands would be useful. Regards, Atis Kind Regards, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 variables are empty.
If al look at the A-number (${CALLERID(num)} this variable is not empty
after the videocall is set up.
Does anybody know how to 'remember' the variable ${CALLERID(dnid)} ?
A global variable is not an option.
Kind Regards.
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[asterisk-users] one CDR instead of multiple CDR
Hi, In my application I jump to different extensions For example: [begin] exten = s,1,Goto(starts,s,1) [start] exten = s,1,Play(welkom) . exten = h,1,Goto(end,s,1) [end] exten = s,1,Macro(end_call) exten = s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not multiple CDR on every call? Kind Regards, image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SET with pipe symbol
Tilghman,
Tx, That was the solution.
Kind Regards,
Arjan Kroon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: dinsdag 29 januari 2008 16:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SET with pipe symbol
On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not last 'options'
entry)
(in my test call ${CALLERID(number) = 061234578)
I tried to escape the pipe symbol by using \ (backslash)
With the same result
Also I tried to place the variable between single or double quotes,
but
with the same result.
Does anybody now how place a pipe symbol in variable.
You can't, in 1.4. This is by design. We have removed this restriction
in
1.6. As a workaround, in 1.4, use the NoOp instruction with the SET
dialplan
function, i.e.
NoOp(${SET(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))})
--
Tilghman
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[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 last 'options'
entry)
(in my test call ${CALLERID(number) = 061234578)
I tried to escape the pipe symbol by using \ (backslash)
With the same result
Also I tried to place the variable between single or double quotes, but
with the same result.
Does anybody now how place a pipe symbol in variable.
Kind Regards,
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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] 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 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. Is there a way to increase this variable? You're going to have to provide more information for us to help you. There are numerous places where the extension string could be getting truncated, so you'll have to look some more: 1) On the console, with verbose set to 3 or higher, when the dialplan is executed, are you showing all of the numbers? 2) If you run wireshark on the wire, does the IAX2 packet show all of the numbers in the CALLED_NUMBER IE? Also, you should know that in trunk, there is a much better way of transmitting independent bits of data about the call, called IAXVARS. We're presently looking at abstracting this into something a bit more protocol independent, but that's the way it is presently. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. Is there a way to increase this variable? Kind regards image001.gif___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
