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Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, mayb...

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'l...

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten =&g...

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

Hi all Is there anyway i could get in the dialplan the amount of time a caller waited in the queue before exiting? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160518/b3b082aa/attachment.html>

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out...

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

...es here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer (terminate) the call. Don't forget AstriCon this year - www.astricon.net On 23 August 2016 at 12:52, Israel Gottlieb <isrlgb at gmail.com> wrote: > You could m and make a moh file that has ringing the first 30 sec and then > the anouncment > > ?????? 22 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > > Thank you for the idea. The problem with RetryDial, is that it...

Hi again! I'm thinking about using my mobile phone to receive (and send) calls when I'm not at home (for example in holiday). I can make my Asterisk reachable from Internet, of course, or I can use a VPN, that's not the problem... My question is: can I log in to the same account from more than one device? If yes, I can just configure my mobile phone with the same login of my

...?????? ? ???: Luca Bertoncello ????: ??? ????, 5 ????? 2015 09:51 ??: Asterisk Users Mailing List - Non-Commercial Discussion ??? ?: Asterisk Users Mailing List - Non-Commercial Discussion ????: Re: [asterisk-users] ?????: Accessing an account from more than one phone Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll...

At the end of the Command you could use options one of them is the c (not apital) which sends a cancel event to the phone http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Fri, Jun 5, 2015 at 9:53 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Zitat von Israel Gottlieb <isrlgb at gmail.com>: > > If you the c option in the dial command it will send answered >> else where sip message to the phone and most ip phones understand that >> The cell will always display a missed call >> > > I'm very sorry, but I can't understand what you m...

...you are dialing a external # then that won't work ? ????? ?????? ? ???: Luca Bertoncello ????: ??? ????, 5 ????? 2015 19:02 ??: asterisk-users at lists.digium.com ??? ?: Asterisk Users Mailing List - Non-Commercial Discussion ????: Re: [asterisk-users] ?????: Missed call Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten =&g...

Hi Guys we have a system that uses a lot of custom hints based on the extension the extensions use the format of ext-system for example 200-pbx01 when starting asterisk the "core show hints" show the correct hints and blf works as expected in the extensions.conf we have _.,hint,Custom:${exten} when running dialplan reload all the hints lose the dashes (-) they become 200pbx01 of course

Hi Does anybody have a idea how I could set sip headers when using call files? I have to call out a specific trunk so I cant use local as the trunk what i'm trying todo is send out calls as "anonymous" but at the itsp it should be filed as being called out thru a specific DID and not the main DID the provider has on file for that I have to send the p-asserted but cant figure out

Hi, I'm trying to add modules compiled from source into a rpm install of asterisk (from digium) on centos and asterisk complains that its not compiled with same options so it won't load it I know I could install the entire thing from source but for other reasons I would like to keep the main things installed from rpm and install whatever else I need from source (or roll my own rpm for

...t;Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive On 02/06/2012 03:19 PM, Paul Belanger wrote: > On 12-02-06 09:15 AM, Jonas Kellens wrote: >> On 02/06/2012 12:25 PM, isrlgb at gmail.com wrote: >>> Your running into a bug and the only way to solve it is to report it >>> and debug it and hope for a fix >>> There is no way someone can help without it being debugged and knowing >>> what's causing it to lockup >>> >>&g...

Hi all asterisk 1.8.23 I have odbc all setup to mysql but cant figure out why the dialplan wont write to the odbc function fubc_odbc.conf [DEVICES] dsn=device-conn ;dsn in res_odbc not odbc.ini readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM call INNER JOIN devices ON call.id = devices.callId WHERE deviceNumber = '${ SQL_ESC(${ARG1})}'

Hi guys i upgraded to asterisk 16.10 and in most recordings you here only leg A in the recording sometimes you might hear a word of leg B Did any body hit this problem? Thanks, israel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200512/90ee8dc2/attachment.html>

?I have the same exact issue. I cannot push any sounds or even Playtones to the caller, unless the channel is answered, which is not possible for billing reasons. I am also using the Local channel & Dial(PJSIP/...). I think this is a bug in Asterisk 13. The Dial function has not answered yet, so the Local channel should be able to play anything to the caller, without answering, in parallel