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My setup : Trixbox 2.6.1 & TE410P running well .: 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and quering it everytime to see the callers

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

Hi List I've a CID lookup hooked onto an inbound route (i m using trixbox) ...it runs well but it returns the value as "CIDNAME<CIDNUMBER>" ... if i just want to display the CIDNAME [leaving the quotes and <CIDNUMBER>] .. how can i do it ? do i have to edit some macro in extensions.conf ? rgds Sriram -------------- next part -------------- An HTML attachment was

Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set "blockcallerid=yes" in zapata.conf ;) Thanks Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I am trying to implement monitoring of asterisk (all 4 spans-i want to show them line by line Up or down) using nagios using below script, but i always get the status as down and red..can anyone let me know how to read an output from nagios plugin ? nagios etc is configured already and is working PATH=/bin:/sbin:/usr/bin:/usr/sbin FAILS="" SPANS=$(asterisk -rnx "pri show

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during "background(menu-filename)" method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting

Hi I;ve trixbox installed with 2 NICs. One NIC carries the MPLS-VPN traffic (only a 1 MB link without internet for carrying voice to another site) while the other NIC has a connection with public IP for internet services on that machine. the first NIC (eth0) has IP of 172.16.0.1 and is connected to router with WAN IP: 10.18.6.254 , the second IP is 203.234.82.98 (eth1). i want to have the

Hi 1. What is the best way to convert wav (44000 Khz) to gsm format for asterisk ? I;ve tried sox command but the outcome is not satisfying...The built-in gsm files shipped with asterisk are simply superb ..How do i create gsm files of similar quality ? Can anyone help me out ? if sox is the only way can anyone tell me the exact command ? 2. Can Freepbx 2.5 installed above asterisk 1.6.0 or

Hi All I am trying to setup : PSTN E1 ---> Asterisk------>Legacy PBX------->Legacy Analog extensions. I've followed steps using : http://www.voipinfo.org/wiki/view/Asterisk-Panasonic i get the green light (sync) on both the 2nd span of digium TE420P (that is cnnected to the legacy pbx pri card) and the pri card of the legacy pbx. but when i try to make a call to asterisk so that

Hi I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls that have duration of 0 seconds, but they are still shown as ANSWERED . how come its possible when duration is 0.00 ? Are the callers billed for such calls ? Rgds Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All am using trixbox with call queues..I've set setinterfacevars=yes in queues.conf and following is dial plan : [test] exten => s,1,Answer() exten => s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten => s,3,Monitor(wav,${FILE_NAME},m) exten => s,4,queue(55365) exten => s,5,Hangup() but MEMBERINTERFACE is always empty - i

Hi All I am getting a strange prblem while installing TE420 on a Dell Poweredge T100 machine. I get a "TE4XXX: version Synchronisation error " nad the machine hangs which needs a hard reboot . Its a new machine and if i install TE410P then installation is successful. Strangely enough i have a working TE420F on another Dell POweredge T100 machine at a different locaiton ... The DELL tech

Hi List Might be a very silly question, I want to make some changes in CDR.C of Asterisk ( i m using trixbox) . I noticed that cdr.c is present inside the main folder of svn asterisk 1.4 branch. If i make any changes in cdr.c how do i update hte changes as i dont see a loadable module cdr.so ? i see rest others like app_cdr.so ,cdr_mysql.so etc .. rgds Sriram -------------- next part

Hi I'm having 2 strange problems (or rather I am doing it wrongly!) ..need some help.I am working on trixbox 2.6 with Asterisk 1.4.22 1. I've a Queue with dynamic agents (ampextensions=deviceanduser mode in amportal.conf), some of agents when they punch in the wrong UserID - they still get logged onto the Queue as a result many of them complain that they don't receive calls on

I've downloaded and installed Trixbox 2.8 (asterisk 1.6) ..I encounter 2 problems for dynamic agents login and logout - 1. When agent from sip phone dials *11 , he is prmpted to enter extension number first - but if he feeds the extension number, asterisk doenst allow him to login but if he enters the AGENT ID - he is given instant access and then he can enter queuenumber* to enter the

Hi I've a queue which has generic zap extensions (of my legacy PBX which is connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx extensions are able to logon to queue perfect.. but Whenever a call comes in queue the status of that extension in "queue show <queuename>" always shows as "NOT IN USE" instead of ringing or In use as shown in a SIP

Hi Everybody I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents

Hi Steve Thanks for all your help, i followed your answers and found on that nagios was being run as user nagios....and if i executed the last command it asked for a password [i tried nagios password,root password etc] but it did not work..it the end i opened nagios.cfg and changed the NAGIOS_USER to root and changed the ownership permissons on the script also to root..I now get the correct

Hi My setup : Trixbox 2.6.1 & TE410P running well ..I've 2 design issues to consider : 1. I need to store the CallerId of the PSTN caller with his language preference so that next time he is played the prompt in his language that he chose the first time.What would be better - storing his number in the Asterisk DB and using Dbput and DBget ? or storing it in MySQL from the dial plan and