similar to: Problem with Callfiles

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

Hello, I'm using Asterisk 1.6.2.0 and I like to call extension #8 from callfile. Unfortunately the #-char ist interpreted just as comment. I got a "Invalid file contents in /var/spool/asterisk/outgoing/callfile, deleting" from asterisk. When I try to escape with \ oder use quotes, I got: \#8,1 failed so falling back to exten 's' or "#8",1 failed so falling back

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

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

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

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 >

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi!I have Samba 3.2.5 as PDC for 20 users with windws XP now I need that 5 users can write into C:\windows folder from each machine in my LAN. I have a Administrators group with RID 544 and i added these users to this group but it doesnt work, I did the same adding to Domain Admin and didnt . There is a something way to give to these users access to can write into this folder Thks.

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer

Hi! I am trying to do a FastAGI server in windows. I am using the example from their page but I dont get anything. Anybody here has experienced with Fastagi in windows and perl that give a rigth direction to do this. I have experience with AGI but fastagi dont *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL

I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or

I installed a queue for a client with 10 officers so far so good. Now the client wants an agent when making a call to this will leave any customer information using the phone as a key. I'm trying to do this app using delphi obviously I will need to connect to the AMI, but do not quite understand how to identify the call you get to a specific agent. Can you give me some point where to start.

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

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

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

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

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

Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [s at from-pstn:1] Answer("DAHDI/7-1", "") in new stack -- Executing [s at from-pstn:2] Playback("DAHDI/7-1", "vm-intro") in new stack -- <DAHDI/7-1> Playing 'vm-intro' (language

Hi! I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I need a lot of script done in perl and I wanna move to windows server. I checked Asterisk::fastagi but I see that everything is for Linux. Somebody has idea to do this in perl. I dont want to change the language. TIA *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase