similar to: cdr call time

Hi Team, i have been facing issues with sangoma card with 16 E1. used LibSS7 asterisk 1.6 with 8 E1 the links are stable , but moment i add another card of 8 E1 for to support 16 E1. link keeps fluctuating any idea why ? Please help Thanks Vinod Dharashive -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi team, I am trying to find solution to hangup b-party call after 1 min with out disconnecting the call of a-party but following dial plan which is disconnect both the calls. Please suggest me the solution. [TB] exten => _X.,1,Wait(${INCOMING_WAIT}) exten =>_X.,2,Verbose(TB) exten =>_X.,3,Answer() exten => _X.,4,Set(mainLoop=0) exten =>

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, 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, 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>

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();

Hi, I am trying to set up asterisk with ss7. Whenever I try to load module chan_dahdi.so, I get the error [Mar 26 17:33:27] ERROR[10437]: chan_dahdi.c:10458 mkintf: Unable to find linkset -1 I have compiled dahdi, libss7, asterisks (am using asterisk 1.6) in that order. Have already set signalling to ss7 in dahdi_channels.conf How do I sort this out? Thanks for your help in advance. Peter.

Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528

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

Hi all, I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of command DIAL when used from the dial plan. Can some one guide me how to get the vaue of

Hi team, Can any one share with me clustering configuration file SS7.conf for single pointcode with four slc. two different machine each host having 2 slc respectively. Thanks Vinod Dharashive Sent from BlackBerry? on Airtel

Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal

Hey everyone ! I have a problem in making outbound calls in PRI connection. I have E1 PRI airtel connection [ India ] [ asterisk-1.2.12.1 on CentOS 4.4 ] zaptel.conf ---------- [channels] language=en usecallerid = yes hidecallerid = no callwaiting=yes threewaycalling = yes usecallingpres=yes transfer = yes echocancel = yes echotraining = yes immediate = no ;group=0 ;context = from-pstn

I'm doing these manipulations on the data frame and wondering why does R have to remember historical data on my operation and not just keep the needed info. Probably a basic fundamentals of the way R handles data .. Pls point me to the manual if possible .. I have this Index data: > head(NIFTY_INDX) Constituents.list.of.S.P.CNX.Nifty X X.1 X.2 X.3

Hi, I'm trying to migrate from Asterisk 1.8 to Asterisk 13 and can't figure this one out. I'm pretty sure the question has been already asked, but I failed to find a solution. Can you modify CDR values in an h-extension? My cdr.conf contains: [general] enable=yes unanswered=yes endbeforehexten=yes initiatedseconds=no batch=no The diaplan contains a simple "h" extension

Dears, I need to save some information on userfield when calls end in Asterisk13, but I have two error cases: 1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but userfield is not set. 2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src e dst correct, and a second line with userfield setting and dst h. I am using cdr_odbc.conf, with Asterisk11.14.0 it

Ok I can't get my CDR values to set from the h extension in either 1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf ; Normally, CDR's are not closed out until after all extensions are finished ; executing. By enabling this option, the CDR will be ended before executing ; the "h" extension so that CDR values such as "end" and "billsec" may

With endbeforehexten=no I actually get two CDR entries. One for the call and a second one for the "h" extension. "","13","10","sip-locals","""13"" <13>","SIP/13-00000006","SIP/10-00000007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be