similar to: call file question

Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBER > > Context: $CONTEXT >

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, whether calling through a callfile or by sending DTMF's. I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are those reliable ways to know when the channel is available for dialing out and the call has been answered?

Hi there, I?ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O?Reilly book by Meggelen, Madsen, & Smith can I find a detailed

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Hello Experts. Im working with Asterisk PBXand freeswitch PBX. I have a challenge with FXO card in Asterisk and i could not solve it yet. I hope you could guide me in this regards. When i want route the call to FXO channels, Before the callee answer the phone (pick up phone), The channel is answered with FXO card. How can change this treat so that the callee dont answer the phone, the channel dont