similar to: With what options is asterisk compiled in rpm's

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, maybe with an example? Thanks Luca Bertoncello (lucabert at

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 create another user. Thanks Luca Bertoncello (lucabert at

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 => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

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>

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 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 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>

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>

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 to

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes 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

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

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

How do I know the options which was compile openldap-servers-2.2.13-4.i386.rpm on my CentOS 4.3 or another rpm? I exactly want to know if the openldap package was compile with -- enable-ppolicy?? Regards Israel

Nice time. # make uninstall ... rm -rf //boot/*xen* ... if somebody use "xen" in kernel name (maybe as suffix), so it will be deleted from /boot/ too. Thanks. Denis.

Hi all, I thought I'd share I script I made (based on some of Leif's works) that lets you download, compile and install Asterisk all in one go; and then removed the dev tools used. We use it quite a bit to provision systems using Ansible, but it is easier than remembering everything every time even if you are using a shell. At the moment I have scripts for Centos 7 and Asterisk 13, but

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, We (NetBSD) just received a bug report that smbclient's mput command usees find(1) with a non-standard option "-maxdepth". This was replaced with a simple call to ls(1). As I don't know how many of the patches from out Packages Collection were sent to you in the past, I'm simply sending you all our patches - please include any you like in future samba releases! BTW,

This is my proposed alternative to the complicated test framework (https://www.redhat.com/archives/libguestfs/2015-August/msg00022.html). In this patch, when we want to run tests on installed packages we just copy in the installed binaries over the source directory, and run the test suite as normal. This is basically the same as the current (not used) contrib/make-check-on-installed.pl script,

emperors1000@aol.com wrote: > > This is being sent on behalf of emperors1000@aol.com > as part of the mailing list that you joined. > List: emperorsclothes > URL: http://www.emperors-clothes.com > ------------------------------------------------------------ > > URL for this article: http://emperors-clothes.com/interviews/dekkers.htm > > Join Emperor's Clothes