Re: [asterisk-users] Hints realtime table structure Ast 11
On 2016-05-18 16:32, Neeraj Chand wrote: Hi All, Has anyone used hints in realtime ? (As in storing and loading hints from odbc) I cannot find a table structure for this anywhere...? Thanks Neeraj Hints are defined in the dialplan so if you are loading your dialplan from a database it is the same thing. I personally use static realtime for my dialplan as I find the "real" realtime scheme very inefficient because you still have to modify the dialplan text file to insert switches (which for me defeats the purpose). -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez dCAP #1349 +52 (55)9116-91161 -- _ -- 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] Hints realtime table structure Ast 11
Hi All, Has anyone used hints in realtime ? (As in storing and loading hints from odbc) I cannot find a table structure for this anywhere...? Thanks Neeraj -- _ -- 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] TDM800 just receive calls, but not make
Hello everyone
I have a TDM 800 on an Ubuntu Server
he's just getting call normally, but when I call any number by this
board, it is silent and not make the call.
look at the log
Executing [629886874@ramais:1] Dial("SIP/2000-000e",
"DAHDI/6-1/29xxx,60,tT") in new stack
[May 18 14:21:31] WARNING[4332][C-000d]: chan_dahdi.c:13433
dahdi_request: Unknown option '-' in '6-1/29886874'
-- Called DAHDI/6-1/29886874
-- DAHDI/6-1 answered SIP/2000-000e
-- Channel DAHDI/6-1 joined 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
-- Channel SIP/2000-000e joined 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
> 0x7f2be401ce80 -- Probation passed - setting RTP source address to
-- Channel SIP/2000-000e left 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
-- Channel DAHDI/6-1 left 'simple_bridge' basic-bridge
<0df64848-6afb-43f3-9f24-5d638aefcb7e>
== Spawn extension (ramais, 629886874, 1) exited non-zero on 'SIP/2000-000e'
-- Hanging up on 'DAHDI/6-1'
-- Hungup 'DAHDI/6-1'
and funny he gives answered right then or wait to ringing.
Out of nowhere he stopped, someone has been there?
Hugs
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Re: [asterisk-users] variable to get waittime of caller exiting queue
thank you
On Wed, May 18, 2016 at 5:48 PM, Faheem Muhammad
wrote:
> Israel,
> You can calculate the time diff by this dialplan snippet.
>
>
> ---
> exten =
> _X.,1,Set(callstarttime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
> exten => _X.,n,Queue(queue1)
> exten =
> _X.,n,Set(callendtime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
> exten =_X.,n,Set(diff=$[${calltime1} -${calltime}])
> exten=_X.,n,NoOp(diff)
>
> -
>
> Regards,
> Muhammad
>
>
> On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb wrote:
>
>> Hi all
>>
>> Is there anyway i could get in the dialplan the amount of time a caller
>> waited in the queue before exiting?
>>
>> Thanks
>>
>> --
>> _
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>>http://www.asterisk.org/hello
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] variable to get waittime of caller exiting queue
Israel,
You can calculate the time diff by this dialplan snippet.
---
exten =
_X.,1,Set(callstarttime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
exten => _X.,n,Queue(queue1)
exten =
_X.,n,Set(callendtime=${STRFTIME(${EPOCH},,%Y%m%d)}${STRFTIME(${EPOCH},,%H%M%S)})
exten =_X.,n,Set(diff=$[${calltime1} -${calltime}])
exten=_X.,n,NoOp(diff)
-
Regards,
Muhammad
On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb wrote:
> Hi all
>
> Is there anyway i could get in the dialplan the amount of time a caller
> waited in the queue before exiting?
>
> 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
>
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[asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?
Hello, I've got the following setup: PSTN ITSP SDSL Modem-Router Gateway - Asterisk with B410P --- SIP Phones Both SDSL Modem-Router and Gateway are managed by my ITSP. Some calls coming from PSTN and forwarded to an other PSTN number have a poor voice quality. How can I best illustrate this ? A friend advised me to simply record incoming DAHDI channel, for instance. How can I then translate record WAV file into meaningful figures ? More generaly, what would you suggest ? Regards -- _ -- 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] variable to get waittime of caller exiting queue
Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? 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
[asterisk-users] Way to replay messages recorded by voicemail()
Hello list,I am asking if "${VM_MESSAGEFILE}" corresponds to /var/spool/asterisk/voicemail/default//INBOX/msg<>.WAVor /var/spool/asterisk/voicemail/default//INBOX/msg<>.txtI would want to replay my message recorded by voicemail() applicationBesides, I would like to replay my message by using this application VoiceMailPlayMsg(${mailbox},msg_id) but i dont know how to grab msg_id in this file (msg<>.txt) without using this file.Is there a voicemail variable which allows us to have this information (msg_id) ?Is there another way to replay the messages recorded by voicemail()?Regards !!!Mamadou NGOMIngénieur Télécommunications & RéseauxMobile: 06 72 45 23 03Skype: Mamadou NumericapNumeriCap – SAS au capital de 30.000,00€ - RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015. siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de Tassigny 83000 Toulon. mail: fina...@numericap.comCentre d’exploitation : « Résidence les Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel :04.42.73.88.52
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