similar to: Asterisk crashed

Hi, Is it working for anyone? I have tried with trustrpid=yes sendrpid=yes/pai but can not get it working, Asterisk cli shows prevented message like this. Connected line update to SIP/1231-00000200 prevented Regards, Zohair Raza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've tested 1.8.6.0, 1.8.4.0 and 1.8.0 I can get proper start and answer time but not the end time of call <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(start) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011-12-16 18:34:48) <SIP/11-00000000>AGI Rx << GET VARIABLE CDR(end) <SIP/11-00000000>AGI Tx >> 200 result=1 (2011 12-16 18:34:48)

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

On 11/24/14 21:17, Mauricio Tavares wrote: > On Mon, Nov 24, 2014 at 9:03 PM, Mark LaPierre <marklapier at gmail.com> wrote: >> Hey All, >> >> Has anyone had trouble with Firefox/Thunderbird? >> >> When I log on if I start Thunderbird first then I can't start Firefox. >> Clicking on a link in an email fails to start Firefox. >> >> If I

Hey All, Has anyone had trouble with Firefox/Thunderbird? When I log on if I start Thunderbird first then I can't start Firefox. Clicking on a link in an email fails to start Firefox. If I start Firefox before Thunderbird then hyperlinks in emails open a page in the existing Firefox instance. Other users on this same machine are having the same problems. This has been happening for that

On 11/28/14 00:47, Philip Manuel wrote: > > On 28/11/14 16:42, Mark LaPierre wrote: >> On 11/24/14 21:17, Mauricio Tavares wrote: >>> On Mon, Nov 24, 2014 at 9:03 PM, Mark LaPierre <marklapier at gmail.com> >>> wrote: >>>> Hey All, >>>> >>>> Has anyone had trouble with Firefox/Thunderbird? >>>> >>>>

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hello, I am getting these errors in /var/log/messages : Oct 12 16:36:15 sambadc samba[7147]: [2013/10/12 16:36:15.817541, 0] ../source4/dsdb/kcc/kcc_periodic.c:664(kccsrv_samba_kcc) Oct 12 16:36:15 sambadc samba[7147]: Calling samba_kcc script Oct 12 16:36:15 sambadc abrt: detected unhandled Python exception in '/usr/local/samba/sbin/samba_kcc' Oct 12 16:36:15 sambadc

Hiii all, I am using asterisk 1.8.9.2 and compile all modules related to calendar. neon version is 0.29.6. OS is ubuntu 11.10. I configured ical for zimbra, caldav for google mail and ews for exchange 2010 calendar. ical and caldav setup working fine and i am getting my calendar events perfectly. But for exchange 2010 calendar i am getting following error. "Unable to communicate with

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

> Nice to know I'm not the only one that tries to use system facilities > the way they were intended to work and has problems. Maybe if you touch > the bug report so they know I'm not the only one the folks will look and > elevate to RH bug just like they used to do? I was surprised that > "crash" didn't get any attention. While only vaguely related to this

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Gotta say, centos has been tough to install and get working. The anaconda installer makes large drive setups horridly tedious (especially if reinstalling a lot). Package selection seems a bit off... I did a virtualization package, no custom..did not install qemu, libvirt, and all attempts to do any virtualization failed due to non-installed packages. Then I decided to select all the

Dear guru's Hopefully someone can shed some light in my issue. I have created a queue with a ringall strategy and all works fine. I want a caller to be able to exit the queue so they can leave a message. I've added the H option in queue command so callers can press * to exit. So far all well, on the cli there is a message the caller pressed * and the extensions stops ringing. But

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

Hi, today I updated the glibc packages on some CentOS machines. After the Update I checked which services/processes I have to restart "yum -C ps" or "needs-restarting" At the most machines I get no information about necessary restarts, but at two machines a long listing : 1 : /sbin/init 386 : /sbin/udevd-d 659 : /sbin/udevd-d 999 : /usr/sbin/vmtoolsd 1103 : auditd 1128 :

I have tried sending this before, but it did not appear to get through. Hello, When using gfs2 with quotas on a SAN that is providing storage to two clustered systems running CentOS6.5, one of the systems can crash. This crash appears to be caused when a user tries to add something to a SAN disk when they have exceeded their quota on that disk. Sometimes a stack trace is produced in

We began experiencing failed vnc connections to the console display on servers that have been updated to EL 6.3. No such failures have occurred on similar connections to EL 6.2 servers. On the client machine a normal vncviewer display appears with the expected graphical login until the mouse pointer is moved within the boundaries of the vncviewer window. At this point the window closes and an

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the