similar to: Idle threads

Hi, I've got different customers that may use the same asterisk. Each user can blind-transfer a call to whatever place they want. But of course the transferring side should be billed for it. What can I do to see the difference between the channels here? If there is an A->B call going on, I'd like to know which side did the transfer - but whichever side does it, I get back to context

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

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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 all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that

I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? The backgrounder behind my question is

I have just installed one of these cards with the intention of using it to send text messages. O2 and Vodafone PAYG SIM cards worked fine (couldn't make calls or send texts before putting on some credit, obviously). Orange and Virgin PAYG SIMs keep showing "Network status: Not Registered". I have not added any credit to these SIMs for fear of it still not working () I

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

I would not have expected this upon implementing sieve. And I really need to get around this limitation. LDA has no problem writing to these large mbox files. Why does sieve have a problem with them? This is very odd. May 10 17:45:04 greer dovecot: deliver(stan): write() failed with mbox file /home/stan/mail/1-Debian-Users: File too large May 10 17:45:04 greer dovecot: deliver(stan): write()

I think I have some failing hard drives, they are disconnected for now. stan {~} root# btrfs filesystem show Label: none uuid: d71404d4-468e-47d5-8f06-3b65fa7776aa Total devices 2 FS bytes used 6.27GB devid 1 size 9.31GB used 8.16GB path /dev/sde6 *** Some devices missing Label: none uuid: b142f575-df1c-4a57-8846-a43b979e2e09 Total devices 8 FS bytes used

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

I've been kind of busy lately, but I wanted to see what's up with ao after the build change. I was able to check out the vorbis module, but when I tried to check out the ao module I saw this: [stan@volsung vorbis]$ cvs -d:pserver:anoncvs@cvs.xiph.org:/usr/local/cvsroot co -r branch_postbeta2 ao cvs server: Updating ao cvs server: Updating ao/doc cvs server: Updating ao/include cvs

I would like to have my asterisk ring my cell phone and let me know when a new voicemail arrives. In fact if it would automatically put into the voicemail menu that would be cool too. In the future I will probably want it to IM me. Are there good examples somewhere of doing stuff automatically on the arrival of a new voicemail ? I noticed a place for the pager email address in voicemail.conf,

Did you mislead me Timo? You said search in 1.1+ is faster than 1.0. I'm seeing approximately 20x *slower* search times in 1.2.10. Via Thunderbird, a full body search of my 11,000+ message IMAP folder hosted by 1.0.15 used to take less than 10 seconds. Since upgrading to 1.2.10 the search is taking over 3 minutes, the imap process servicing the client pegging one CPU at 100% for the

Hi, I'm not sure if this is some throttling/DoS protection or a bug. I didn't see notice anything like it mentioned looking at the NEWS file from hg, this is on 2.1.7 on Debian stable. I have a client doing IDLE on INBOX.vomiteer. When individual messages arrive with some time interval in between them, I get EXISTS events for each message as expected. However, when I store a bunch of

Hi, I've been noticing a lot of these messages lately: "NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?" Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots