similar to: Realtime and Local Channel Crash Problem 1.8.3-rc2

Hi, Have just been testing asterisk 1.8.0, 1.8.0-rc5 and a trunk version from about half an hour ago. IAX Friends (Zoiper Softphones) don't stay registered for more than a few seconds they start out with status unknown and quickly become unreachable, I am using realtime with postgresql, with tables and configuration that have worked fine for IAX in 1.6 and a trunk release from a few months

Hi, I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand (&) to combine multiple days / day ranges but this gives me an error. For example mon&wed gives the error (in the asterisk console): [Nov 23 18:04:27]

New problem, I figured out how to get the extension working and internally it works just fine. If I pick up a phone and hit 501 my cell starts ringing. However if an inbound caller dials that extension Everything seems to stop when it trys to bridge the two trunks together. Sound familiar to anyone? exten => 501,1,Macro(dialout-trunk,1,5551212) exten => 501,2,Wait,1 exten =>

CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans"

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hi, I have not esatblished a working rsync config on my server, but have found one major problem in which i hope there is a solution to. I'm based in Norway, with scandinavian clients. Of course we use several "special characters" in our keymap, which is not supported in US keyboards etc. I know there is a better word for this, unfortunatly i can't remeber it, so i hope you

Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right

I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten=> 501,1,Congestion() exten=> 501,2,Hangup() exten =>h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =>h,2,Hangup() With the above, the call comes into the trigger number, then the call

Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the

On Fri, 2015-06-26 at 09:40 -0600, Shawn Heisey wrote: > On 6/26/2015 5:48 AM, Andrew Beverley wrote: > > I'm configuring some Sieve scripts for virtual users. I'd like to keep > > the Sieve scripts somewhere "sensible". > > > > Currently, all the mail goes into /var/mail/vhosts/<domain>/<mailbox> > > > > So I thought a good

Hi, I have been experimenting with CEL in a trunk version of asterisk for some time and have upgraded my test machine to 1.8.0 today. Made a few calls and it looks like the eventtype field is missing in the CEL insert query when using ODBC. I see the following errors on the console: [Oct 22 21:46:09] WARNING[952]: res_odbc.c:634 ast_odbc_prepare_and_execute: SQL Execute returned an error -1:

I'm trying to dial using this script in perl: (asterisk_dial.pl) --- #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $pin = $AGI->get_data("beep", "10000", "3"); chomp($pin); $AGI->exec('SetVar',"NUMERO=$pin"); exit(0); --- This script sets 'NUMERO' with any value... In my

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hi, I'm going to have a go with realtime mysql. Just wondering, most examples i came across while googling, was with 1.6 systems. So any drastic changes with 1.8.3, table-layout? other pitfalls? hw

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I