[asterisk-users] Need some info on cmd Bridge (Confbridge)
Hello, Perhaps i'm wrong but i don't find a real documentation for cmd Bridge (i've take a look to the source code but i'm not a guru and i probabely miss something): Is it possible as for cmd meetme to have a context to return on 'exit'/end of the bridge? (in fact i think 'no') I've done a workaround with meetme but i would use Bridge (or confbridge) if possible (perhaps a channel redirect?) Thanks a lot for any suggestion Best regards D Envoyé depuis Ma Messagerie SFR. 10 Go de stockage - en savoir plus.-- _ -- 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] Stupid question: Why Cmd Dial and Queue haven't same options?
I apologize, my English isn't better than the last year So, - why Queue has some options like 'caller can continue in his dialplan' (Dial has g and F options for callee/caller) and not he same option for 'callee' ? - no option in queue to send message (for only calle/caller) when bridged as Dial has - idem to announce periodically 'X minutes before hangup' (in Dial, not in Queue) Probabely, i've missed something. perhaps some workaround ? Thank's a lot for your attention -- _ -- 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] Stupid question: Why Cmd Dial and Queuehaven'tsame options?
OK Danny, but of course, i don't mean 'exactely the same options'.hum i don't really understand why it's possible for the caller to continue in his dialplan (= cmd Dial option g) and it's not possiblef or the callee (as option 'F transering' on DIAL). Reading the source code in app_queue.c, i think it's possible to initiate/monitor some action after caller/callee are bridged (and some comment on the beginning of this app: Optional monitoring of calls, started when call is answered), i'm wrong ? Please, could you be more specific about 'item 3' ? some 'barge in' action with AGI ? i'm Beotian (sorry for grek people) One more time best regards - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, February 16, 2010 5:20 PM Subject: Re: [asterisk-users] Stupid question: Why Cmd Dial and Queuehaven'tsame options? callee in queue-land is always an agent who picks up the call instead of getting it transferred. In some concepts, queue is more like a conference than a transferred/dialed call. Item 3 could be accomplished with an AGI/AMI command. -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of didier.cuffaut Sent: Tuesday, February 16, 2010 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Stupid question: Why Cmd Dial and Queue haven'tsame options? I apologize, my English isn't better than the last year So, - why Queue has some options like 'caller can continue in his dialplan' (Dial has g and F options for callee/caller) and not he same option for 'callee' ? - no option in queue to send message (for only calle/caller) when bridged as Dial has - idem to announce periodically 'X minutes before hangup' (in Dial, not in Queue) Probabely, i've missed something. perhaps some workaround ? Thank's a lot for your attention -- -- _ -- 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] How to exchange/get $variables from/to each channel on cmd Dial
I apologize for my poor English. So, i don't really understand 'how to' realize thus When you use the cmd Dial and want to get $ from caller channel to callee (or callee channel from caller), which way is the right way ? Sorry, i've take a look to the wiki and asterisk code and is'nt limpid (for me) use Macro in the dial cmd hum it's on the caller side and i doubt use local channel ? nor more explicit. something like Importvar ou shared(var) ? not really evident to use because documentation is 'abscons' (for me,sorry,the wiki is very light on theese 2 features)) at least, i need some examples (not in the voip wiki and no more on * doc, afaik) another way ? Thanks for any help or reference. Happy Christmas and 2010 for everyone ___ -- 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] DTMF
- Original Message - From: Jeff LaCoursiere j...@jeff.net To: asterisk-users@lists.digium.com Sent: Thursday, April 16, 2009 5:25 PM Subject: Re: [asterisk-users] DTMF Hmm, let me rephrase that (now that I have googled a bit). I am having trouble with DTMF tones over two IAX trunks: Polycom501---ast---[IAX]---ast2---[IAX]---provider Both IAX trunks were ulaw, and that worked fine. I recently changed the first leg to be g729 (as their internet connection is lower bandwidth). Now DTMF doesn't seem to pass. In my searches just now I see that dtmfmode is not actually a valid keyword in iax.conf. So may I assume that dtmfmode is inband only over IAX (since adding compression seems to have killed it?). That would suck. j On Thu, 16 Apr 2009, Jeff LaCoursiere wrote: Is there a way to show the negotiated DTMF mode for an active channel? Cheers, 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF
- Original Message - From: Jason Aarons (US) To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, May 22, 2009 10:32 PM Subject: Re: [asterisk-users] DTMF Then if it's a IP interface (SIP, etc) have you tried a sniffer trace (wireshark, etc) to verify the packets are being sent correctly to carrier? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, May 22, 2009 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF Is this inbound calls to your automated attendant? Or Outbound calls to say a bank ivr out in the pstn? What direction? What is your interface/carrier? T1, SIP, H32? And what method are you using for DTMF? Eg inband, out of band, what rfc, etc? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Friday, May 22, 2009 3:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF We are facing alot of problem in the DTMF. At times we are unable to do the verification because whenever we press the numbers for verification it does not detects and at times it detects the wrong number for instance if the customer is having the phone no. as 1234567890 it will detect 123467890 or 234567890 . And we also face the problem that the line get disconnected while doing the verification and also at times the conference is not working . -- Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- ___ -- 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] [FIXED] Re: call-limit on a per destination basis
Ne manque t il pas des espaces entre } 24]
- Original Message -
From: Jean-Michel Hiver
To: Asterisk Users Mailing List - Non-Commercial Discussion ;
klaus.mailingli...@pernau.at
Sent: Friday, February 27, 2009 2:01 PM
Subject: [asterisk-users] [FIXED] Re: call-limit on a per destination basis
The correct syntax for GotoIf is:
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}24]?500)
Otherwise it seems to evaluate the string number 24 which is always true.
Duh...
Thx
JM
--
Jean-Michel Hiver - Synapse co-founder CTO
GSM +262 692 828 070
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Re: [asterisk-users] Wanted information
Sorry, I agree with Dean Collins and David BUT: is it very friendly to said 'Cut this etc etc' - these comments also use my bandwith. i'm loosing some time reading theese posts +1 +2. why not + 99 ? one shot and be quiet... thank's a lot (hum. me too, i apologize, but too much comments on the list..) - Original Message - From: Dean Collins d...@cognation.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 29, 2009 1:48 PM Subject: Re: [asterisk-users] Wanted information Ambarish, no you cannot install it on a PC running windows XP, it' needs it's own dedicated server. You can run it from a virtual machine for testing/learning purposes but for real world implementation it will need it's own computer. Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of ambarish.deshm...@wipro.com Sent: Thursday, 29 January 2009 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Wanted information Hi, Ambarish here from India, New (beginner) to asterisk here, Wanted to know how can I install asterisk on Windows XP SP2, with AMD Athlon 64 processor 2.20 Ghz, 512 RAM Can anybody help / guide me in this? Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] Call file in the future
First, thanks for your help
Ok, i going to do a script and call ot with only one 'System' (cf Gordon
Henderson) and take a look to 'incron' (T Cohen)
Just need some explanations:
1) If the call file 'failed', an 'exitstatus' is happendGood
How to check/get these $ and put in in an * $ ? (of course, the call file have
to have archive= yes and go to 'outgoing-done')
sorry, i'm not a linux guru and it's not a pure Asterisk pb. Anyway, could
someone show me the complete exact way and syntax to do this?
Using something as: $ egrep -vw (^#|^) file | awk -F '{ print $2 }'
(or some use of awk)
2) From my first post, are these lines OK or wrong? (syntax error?)
tmsp = the delay in future.. say 100 seconds
exten= ra,n,System(NOW='date %S')
exten= ra,n,System(let NOW=$NOW+$tmsp)
exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1
+%Y%m%d%H%M. %S)NOTE THE 'M. %S'
*
or this way ?
exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])
exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)
*
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[asterisk-users] Call file in the future
Hello,
I read a thread on the asterisk dev list (call file handling suggestion)
May i have some comment/opinion on these two ways below to place a call file in
the future ? (from the wiki and the asterisk book but added typos and stupidity
come from me)
The best is ? (and should work ?)
tmsp = the delay in future.. say 100 seconds
exten= ra,n,System(NOW='date %S')
exten= ra,n,System(let NOW=$NOW+$tmsp)
exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1
+%Y%m%d%H%M. %S)
*
or this way ?
exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])
exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)
*
next step:
exten= ra,n,System(touch -t $TOUCH_TMSP /tmp/${idclient}.call))
exten= ra,n,System(mv /tmp/${idclient}.call /var/spool/asterisk/outgoing)
Thanks for your attention, happy 2009. and perhaps a reply ?
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Re: [asterisk-users] Dial command and its g option
Hi, With the g option, you just have to continue in the CALLER Dialplan, you have nothing to do, just continue your Dialplan i.e: exten= s,n,Dial(what you want) = and when the Called hangup you're goto the next line exten= s,n,Goto(where you want) or exten= s,n, 'DO WHAT YOU WANT: playback, background and so' After the CALLED party hangup (of course, not the caller), the CALLER continue in his dialplan.. Hope i'm not misunderstanding your question.. BUT if the two legs hangup, you have to use DEADAGI on the h extension.. - Original Message - From: voip crazy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 12, 2008 12:25 PM Subject: [asterisk-users] Dial command and its g option I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t how its works, someone could put a example about the way to use it. Thanks you in advance. VoipCrazy ___ -- 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] Cmd Dial (a group) and 1) who pick up the call 2) How to use the G option
Hello, 1) The goal is to store the id of the operator who take the call in a mysql database andI don't know the (best) way to know which device take the call when we do a Dial cmd to a group of phones Some $var as DIALEDPEERNUMBER ? some inheritance ? using the G extension ? (and how to?) Thanks a lot for any reference/links or example 2) In the same idea, i don't mean exactely what is it possible to do with the G option of Dial cmd Ok the wiki is clear...callee/caller go to the same extension with priority x and x+1.. Sure, i'm not inventive.. what can i do with that ? are the $var of each caller/callee channel accessibles from the other leg (example: transfer the $id of the callee channel to a caller channel $var) ? And when a leg hangup, the other could continue on dialplan (as g option for the caller)? More: could the two legs do separate action in the dialplan during the call ? If someone have some example of use (or links etc) Have a nice week end ___ -- 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
