similar to: 'h' extension on call-out

Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized

Hi all! Is anyone using the $subject setup? What I would like to do the following setup: 1. OXE is setup for receiving calls, handling Agents 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI The incoming calling route: 1. OXE handles incoming calls, answer 2. Transfer to extension 9xxx 3. Asterisk answer (using one channel) 4. IVR is handling calls 5. If needed IVR

Hello! I am using Asterisk as transparent voice recorder for calls (isdn <-> asterisk <-> pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both

Hello! I would like to setup an NTP server for my Windows network using CentOS 6.3 with firewall turned on. As I learned the NTP protocol uses port 123 UDP. I have two NIC cards. One for internal network and one for access internet. Both cards in private address range. The problem is when I am using firewall described below the client cannot access the server. No idea why. Without firewall

After more investigation, I decided to just recompile asterisk (on my newly upgraded Fedora core 4 system). Make dies with this error: "No rule to make target 'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h" It seems this directory is gone under FC4, and replaced by No rule to make target 'usr/lib/gcc/i386-redhat-linux/4.0.0/include/ I can't find the

Hi all! On outgoing call sometimes Asterisk use/give back the caller id sent back by called number instead of number called by me. This is annoying and misleading statistics if other side use some exotic number. For example I have called number 12345678 and CDR include the number 333 as callerid which was sent back by called number/set/switch/whatever. Normally it cannot be an issue but I have

Thanks Dovid! Indeed looks a bug but regardless of this, this problem made me think that the HANGUPCAUSE could be used for this purpose with benefits. I couldn't find an explanation about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better

Hi, I searched both AsteriskWin32 0.66b build from Asterisk 1.2.26.2 and AsteriskWin32 0.60b build from Asterisk 1.2.14 for Meetme application. But both are showing "The application is not registered" and that module (app_meetme.so) is also not present.Plz anybody can tell me the reason for this. preethy ** -------------- next part -------------- An HTML attachment was scrubbed...

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all! I have the following setup: Phone lines -> traditional PBX -> Welltech 3802 -> VPN -> Asterisk -> Linksys PAP2/Welltech ATA-151 -> phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup

Hi all! I would like to make a service with recording sounds and playing back to caller. I had wrote the script but it failed at Read statement with file not found error. I have put some file test into script and this is what happen on verbose level 9. -- Executing [8298 at default:8] Record("DAHDI/27-1",

Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

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

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Thanks for the feedback. I do agree with having multiple smaller servers. When I was first approached with this task I mentioned as much. However, the current desire is to work with already existing hardware. That is out of my hands at the moment unless it just can't be done. I will explore Freeswitch a bit soon to compare it as well. I am struggling to find what the bottle neck is in