similar to: Get calling channel before pickup

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

Dear R community, I am using R 2.1.1 on Windows XP, package lattice Version 0.12-5, and want to add text (sort of a dat-stamp actually) to the lower left corner of a lattice plot, prefarably _after_ the plot has been created. Here is a simple example what I do in base graphics: # base graphics: > plot(rnorm(100), rnorm(100)) > mtext(as.character(Sys.Date()), side = 1,line = -2, outer = T,

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

What sound cards are recognized by Centros? I have installed it on an old 400MHz IBM Aptiva that has Crystal Audio built-in. However it does not seem to recognize that, nor a Sound Blaster, nor a Sound Blaster 16 that I have tried. Any cheapo cards that it would recognize? Mike

Max H. wrote: I totaly agree with you, and I wanted to tell especialy to M.Hockings that I am sorry if my answer appeared to be a little agressive, it was not that way I wanted to tell... Linux is freedom and should stay like it... Use it the way you want... of course, I think the problem with your sound card could be solved with isapnp, because those sound cards (on older IBM Aptiva) were often

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

Hello list, is it possible to set a variable (channel variable) from within another channel ?! I'm currently working with 2 channels that I bridge afterwards. It would be good to set a variable in one channel when something occurs in the other channel. If some variable is not set in channel 1, then this means something for channel 2. But from within channel 2 I can not see the variables

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

Is there a way to hang up on a sip channel. One of my phones is saying it's busy while it's not (even after rebooting it). I logged into asterisk, and did a sip show channel 232, and sure enough it thinks it's on a call. How can I force it to close?

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790]

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

Hi there, I got a problem when trying to read in a .cel file using ReadAffy(). R codes: require(affy) ReadAffy(filenames="CH1.CEL") It failed and I got the error, Error in read.celfile.header(as.character(filenames[[1]])) : Is CH1.CEL really a CEL file? tried reading as text, gzipped text, binary, gzipped binary, command console and gzipped command console formats Also, I tried

I'm having some trouble getting the channel status with an AGI script. #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->channel_status('Zap/1-1'); I am now stuck, and don't know how to get the return codes: -1 There is no channel that matches the given <channelname> 0 Channel is down and available 1 Channel

After configuring my home network, aMSN segfaults. I posted this issue originally in the aMSN forums under this thread: http://www.amsn-project.net/forums/viewtopic.php?t=7593 I was told that my issue is related to SAMBA, referring this thread: http://www.amsn-project.net/forums/viewtopic.php?t=6343 After uninstalling SAMBA, aMSN stops segfaulting and works as expected. After installing it