similar to: combining AGI with dialplans

Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten

Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks...

Hi, I would like to know how to see which channels are used in my PRI E1 link from Asterisk to another locally-connected commercial PBX. If I run "dahdi show channels", I can see the used channels in the second column "extension" but only if it's an "incoming" call (ie. legacy PBX to Asterisk). If I dial from an Asterisk extension to an extension in the other

Hey all, There's a website I help run that uses the Cassandra DB as its database. I notice that if I run the web server in SELinux permissive mode, the site works fine. But if I put it into enforcing mode, the site goes down with this error: Warning: require_once(/McFrazier/PhpBinaryCql/CqlClient.php): failed to open stream: Permission denied in

Hi, I wanted to use the same extensions for Pausing and UnPausing queue members. Is that a variable that is set up with the agent status (on call, available, not logged, paused) so that I could use it to make some logic in this extension? exten => 111,1,Set(AGENTEPARADESLOGAR=${$[AGENTBYCALLERID_${CALLERIDNUM}]}) exten => 111,2,PauseQueueMember(|Agent/${AGENTEPARADESLOGAR}) exten =>

I had to switch quickly to 1.4.9 from1.2.4 and now I can only get 2 digits into the dialplan. error -- Invalid extension '81' in context 'impact' on SIP/207.174.111.34-b77167f8 I pressed 8107 and ideas my dial plan is (part of it) [impact] exten=>s,1,Answer() exten=>s,n,Set(CALLERID(name)=Impact - ${CALLERID(number)}) exten=>s,n,Background(IMPACT)

I''m working on porting a dynamic website done in PHP over to mod_ruby. I have everything set up and working nicely (I can make a .rhtml page, put ruby code between <% %> and it works etc), but when I try to use "Eruby.import" as a sort of analog to PHP''s "require_once()", I run into trouble. 1) eruby is in my cgi-bin 2) here''s a sample of

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

Hi Jeremy, An easy way to start troubleshooting these is to look at the audit logs and > see what SELInux is blocking. You have /McFrazier in the email.. if that's > off the root tree than unless you've set permissions to allow httpd to look > at tat folder, I bet that's one problem. > if you run ls -Z you can see the labels that are present on those folders, > that

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so

Hi All, i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call. But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on

Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be called 'callend': $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstr"); $answeredtime =

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

I am curious partly because it has occurred randomly in my asterisk system. How does one go about creating a .call file for placing a call between two extensions/phones? I know this has been mentioned and is probably in one of the wikis somewhere, but I am unsure exactally how to go about doing it. Can anyone point me in the right direction. Dan

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

Hello, How do I install phpagi? http://phpagi.sourceforge.net/ I couldn't find any info about setup in that site, and I couldn't email the developers.so I'm lost. I know it isn't a real question for this list, but I suppose many people here already have installed it. So, how can I install it? Thanks Carlos -------------- next part -------------- An HTML

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using