Re: [asterisk-users] Impromptu conferencing
also sprach Brandon B. bran...@brellsystems.com [2012.12.03.0132 +0100]:
[all-inbound-for-999]
; inbound extension through a conference room
exten = 999,1,MeetMeCount(999,COUNT-999);
exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
exten = 999,3,Dial(SIP/99,999,G(6));
exten = 999,4,Hangup;
exten = 999,6,MeetMe(999,FAqx);
exten = 999,7,MeetMe(999,Fqx);
; bypass the conference room for multiple inbound calls
exten = 999,10,Dial(SIP/999);
This is an interesting approach, but I am still not sure how to add
the third party. Sure, I can call them up and tell them to dial
a number, but I'd really rather be able to just switch them in.
What would need to be done for a user to e.g. suspend the
conference, dial another number and finally merge the channels? Do
I need the manager API for that, like this:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#Mergingconferences
?
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[asterisk-users] Calling from SIP client then bridge between two end points
Hi All; How I can acheive the following: From sip client softphone (from the iPhone for example), if I dialed a number that I need to call it, then a call to be initated to a specific number through DAHDI channel and another call to be initiated for the destination number (the number that I dialed it from the softphone) and these two calls to be linked togethor (to call each other directly). So the call from the softphone just to important in the begining to trigger this scenario. How this settings to be done? Regards Bilal -- _ -- 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] Calling from SIP client then bridge between two end points
Assuming a sip client lands calls in a context named 'from-sip', I would
have following sort of logic in dialplan.
[from-sip]
exten = _X.,1,Noop(${EXTEN},${UNIQUEID})
same = n,System(echo -e
Channel:Local/s@specific-number/n\\nContext:external\\nExtension:s\\npriority:1\\nSet:
external_num=${EXTEN} /tmp/${UNIQUEID}.call)
same = n,System(/bin/mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing/)
same = n,Hangup()
[specific-number]
exten = s,1,Noop()
same = n,Dial(DAHDI/g0/XX,30)
[external]
exten = s,1,Noop()
same = n,Dial(DAHDI/g0/${external_num},30)
You can also use Asterisk application 'originate' in place of callfiles. I
normally prefer local channels in Callfiles or Originate so that I can have
better call control through dialplan.
--Satish Barot
On Mon, Dec 3, 2012 at 3:08 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
How I can acheive the following:
From sip client softphone (from the iPhone for example), if I dialed a
number that I need to call it, then a call to be initated to a specific
number through DAHDI channel and another call to be initiated for the
destination number (the number that I dialed it from the softphone) and
these two calls to be linked togethor (to call each other directly). So the
call from the softphone just to important in the begining to trigger this
scenario.
How this settings to be done?
Regards
Bilal
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Re: [asterisk-users] setvar from chan_dahdi.conf
You wouldn’t do it from chan_dahdi. You would set the context in chan_dahdi, then the context would set the variable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chet W. Stevens Sent: Saturday, December 01, 2012 9:05 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] setvar from chan_dahdi.conf Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example: [trunkgroups] [channels] [phone_template](!) usecallerid = yes hidecallerid = no callwaiting = no threewaycalling = yes transfer = yes echocancel = yes echotraining = yes immediate = no context = longdistance signalling = fxo_ks [test1](phone_template) callerid = Test 1 (111)222- setvar=myvariable=test dahdichan = 1 I have tried every example I have been able to find but nothing appears in a DumpChan. Thank you. Chet Stevens -- _ -- 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] Query list of defined channel variables via AMI
Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get. -- _ -- 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] Query list of defined channel variables via AMI
Check DumpChan http://www.voip-info.org/wiki/view/Asterisk+cmd+DumpChan http://wikiasterisk.com/index.php/Aplicaciones_B%C3%A1sicas#Aplicaci.C3.B3n_DumpChan Regards, Rafael Rincón IP-COM, Inc Senior Network Engineer rrin...@ipcomnetwork.com 3100 SW 145th Ave. Suite 410 Miramar, FL 33027 +1 (305) 477 2902 Miami x 111 +1 (877) 55 IPCOM US Toll Free x 111 +52 (55) 3692 4266 Mexico City x 111 + 57 (1) 742-3408 Bogota, Colombia x 111 CONFIDENTIALITY NOTICE The information contained in this email is intended only for the individual or entity to whom it is addressed. It may contain confidential and privileged information and if you are not an intended recipient, you must not copy, distribute or take any action in reliance upon it. If you believe you have received the email in error or doubt the authenticity of email apparently from this source, please notify the sender. You should then destroy and delete the message from your computer. On Dec 3, 2012, at 10:17 AM, Alex Villací s Lasso wrote: Is there a way to list the names of the channel variables that are currently defined on a given channel via AMI? I know of GetVar and SetVar, but GetVar needs the name of the variable to get. -- _ -- 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] Impromptu conferencing
On Mon, Dec 3, 2012 at 1:09 AM, martin f krafft madd...@madduck.net wrote:
also sprach Brandon B. bran...@brellsystems.com [2012.12.03.0132 +0100]:
[all-inbound-for-999]
; inbound extension through a conference room
exten = 999,1,MeetMeCount(999,COUNT-999);
exten = 999,2,GotoIf($[${COUNT-999}=1]?10);
exten = 999,3,Dial(SIP/99,999,G(6));
exten = 999,4,Hangup;
exten = 999,6,MeetMe(999,FAqx);
exten = 999,7,MeetMe(999,Fqx);
; bypass the conference room for multiple inbound calls
exten = 999,10,Dial(SIP/999);
This is an interesting approach, but I am still not sure how to add
the third party. Sure, I can call them up and tell them to dial
a number, but I'd really rather be able to just switch them in.
The third party has to dial into the system at some point. You can only
switch in existing calls, so your idea is nonsense to me because you
haven't specified how else the third party is in or gets a channel into the
system.
The simplest solution would be to have the third party dial into or be
connected into conference room 999. The first two parties are already in
the conference. The conference room 999 could be available by dialling 8999:
[call-to-conference-room-999]
exten = 8999,1,MeetMeCount(999,COUNT-999);
exten = 8999,2,GotoIf($[${COUNT-999}=0]?10);
exten = 8999,3,MeetMe(999,Fqx);
exten = 8999,10,Hangup;
Anybody who can dial 8999 in the above context will join the conference.
What would need to be done for a user to e.g. suspend the
conference, dial another number and finally merge the channels?
Suspending the conference would mean that phone 999 puts the call on hold,
and then does one of the following: (a) dials a third party and then
transfers the connected third party call into the conference room, (b) uses
the conference feature of their phone to connect a third party to their
channel already in the conference room or (c) invites a third party to dial
a number which connects them to conference room 999.
Do I need the manager API for that, like this:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#Mergingconferences
Sure, that will work in theory.
Look into the Bridge command which is available inside a dialplan using
extensions.conf. You can take a call which will end up in a conference room
and before that call joins the conference room it will find* other calls in
the system meant to be in the conference room, bridge those other calls
into a conference room, and finally join the conference room. In this case
the third party call initiates the conference. Alternatively, in the
example above, the conference room always is used so the third party just
needs to join it.
Good Luck.
* finding calls is not possible unless you make them findable
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[asterisk-users] How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
If you are trying to provide CDR files to a billing service, such as WebCDR.com, you need to provide files containing your latest call data every 15 minutes or so. I wrote a script and a cron job that will create a new CDR file every 15 minutes with the latest CDR records, without interrupting call flow. You do not need to make any changes to your Asterisk configuration to use these scripts. There are two files that you need to install on your Asterisk server: asterisk-cdr-rollover.sh – A bash shell script. Copy this file into /usr/local/sbin. This script moves the file /var/log/asterisk/cdr-csv/Master.csv to a new file named /var/log/asterisk/cdr-csv/cdr-MMDDHHMISS.csv, where MMDDHHMISS is the current time. A new zero-byte Master.csv file is created using the default umask of the user running the asterisk process. Asterisk will start writing to the new Master.csv file at the end of the next call. asterisk-cdr-rollover – This is a cron job. Copy it into /etc/cron.d and it will run the /usr/local/sbin/asterisk-cdr-rollover.sh script once every 15 minutes. The cron job is set up to run as the user “asterisk”. If you are running asterisk as “root” or some other user name, edit the asterisk-cdr-rollover cron job and change the name of the user running the script to the same name as the user running the asterisk process. The latest versions of these two files can be downloaded from GitHub: https://github.com/earlruby/asterisk-cdr-rollover. -- _ -- 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
