Im trying to set a CDR userfield to a custom value. This value may contain a | but its really just part of the value. However, Asterisk keeps warning me about the application delimiter not being a pipe. Its NOT an application delimiter (its just p..
This morning someone tried to make sip call through my Asterisk. My server just drop these calls and record them in CDR with IP address: 2012-11-28 06:30:51 SIP/216… 1000 1000 hangup 999011972592249388 ANSWERED 00:01 Hacker: 220.127.116.112. 2012-11..
Im trying to understand how AGI works. Im using php-agi library from sf.net.*CLI> agi show commandsYes exec Executes a given Application*CLI> core show application setMy PHP-AGI script contains:$AGI->exec(Set, CUSTOM_VAR=2);$AGI->exec(NoOp, \DEBU..
Hello;I remember that I saw at Asterisk website (this was maybe before 1 year or around) some pages are talking about having SDK and APIs for asterisk that will be used to build softphone for mobile and will be used to build some applications for asteri..
I have 2 analog trunks.They answer the incoming call, do the welcome message, ask for the extension, when a valid extension is entered it rings the right SIP phone BUTwhen the SIP phone is answered, the SIP phone keeps ringing and the call is not connect..
How consistent has the syntax for extensions.ael been from version to version?extensions.conf has annoyed me in this regard.i.e.: commas to pipes, pipes back to commas, macro to gosub..
Can anyone help me find a specification for the 23B + D Asterisk TDMoE(packetized PRI) link? I am having occasional packets arrive from the Asterisk Server that do not have the standard four leading characters in front of the MAC addresses, and am ..
at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information.Is there a way to exclude certain queues from being logged into the queue log ?Than..
members, Thanks for the support so far as I try to install and test my first Asterisk system. I was able to finally install asterisk-18.104.22.168 with libpri-1.4.13 and dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in the online documentat..
,I face the following problem on incoming calls from my provider which uses Asterisk 22.214.171.124, our asterisk being 126.96.36.199. Incoming calls are not sended to the context set in provider sip.conf definition, but are going to the default context set..