similar to: Re: Marco and Realtime Extension Problem [SOLVED]

Dear All, I have a problem with the Marco and the Realtime Extensions in my extensions.conf. The problem is that when I exit from my Marco, I should return to my calling context, which is default but the next step for it should be switch statement which will use realtime extension. Somehow I am getting the following error below with autofallthrough=yes : -- Executing

Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon,

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

FYI, just pushed: >From 322ff984a39d53422b772bfeb8f69e7c648da8c0 Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Fri, 14 Aug 2009 21:01:48 +0200 Subject: [PATCH libguestfs] build: avoid "make sytnax-check" failure * daemon/configure.ac: Change a leading TAB to 8 spaces. --- daemon/configure.ac | 2 +- 1 files changed, 1 insertions(+), 1

I apologise for writing again about the problem with using stepAIC + multinom, but I think the reason why I had it in the first place is perhaps there may be a bug in either stepAIC or multinom. Just to repeat the problem, I have 126 variables and 99 cases. I don't know if the large number of variables could be the problem. Of couse the reason for doing a stepwise method is to reduce this

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I''d like to use :through to create a web of associations like: class Thing < ActiveRecord::Base has_many :child_things, :through => :thing_thing has_many :parent_things, :through => :thing_thing, :some_other_option? end class ThingThing < ActiveRecord::Base belongs_to :thing belongs_to :child_thing, :class_name => ''Thing'', :foreign_key =>

Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup

Hi, sorry to bother you guys with a simple sytnax question; i have a loop of objects taking place (ie, for page in @pages....xxxxxxx....end) and a link associated to each pages so that in the end it looks like this: page1 (link) page2 (link) page3 (link) . . . page n (link) (all of this done by putting a simple ''link to'' in the for loop.) now i need to seperate two pages

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

Dear All, I am having this strange problem on my Asterisk 1.2.1. We have a web dialer that can register to the Asterisk box in Hong Kong, but another user using the same account can't register to the Asterisk box using the same web dialer. Below is an output of the sip debug logs. It seems that the digest is missing the username and password, but why? I have also have this call flow for

Hi, I have already set canreinvite=no in the sip.conf and also used the NAT=yes. But the funny thing that was in one case the user call and it wasn't working (one way audio as described) using an online dialer and then tried again using X-lite it was working. Then hanged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online

Dear All, I have a very strange situation here and wondering if anyone can assist me. I am trying to connect an H323 call from an GnuGK to Asterisk 1.2.1 which routes the call to an SIP Hard Phone. The funny thing that I can collect the connect but the call always drop about 1 second or 2 seconds after it is connect. I am not sure if this will help but I do see some 'Trapped RCF' in

Hi All, I am having some issues with Asterisk 1.8.3 extensions with a SIP Phone and an gateway. My setup is that I have my SIP Phone setup to register with the gateway. Then the gateway should sent calls to the Asterisk as a type of friend. This works fine if the SIP Phone configuration username and password isn't already set into the asterisk. The configuration of the SIP Phone username

It seems that the UPS I wrote in my previous mail uses the Megatec protocol, but does not support the S<n>R<m> "Shut Down and Restore" Command. This means that the server will shut down cleanly, but it will not boot automatically when the utility power comes back. There is another shutdown command according to the protocol description, although I do not really understand the

Define non-Voice T1 porject? You do know that TDMoE does not travel over long distances, You can not route or otherwise take it off of a single ethernet segment. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Mike Hammett > Sent: Thursday, February 09, 2006 1:20 AM > To:

Hello all, I'm running Asterisk 1.4.5 and Zaptel 1.4.3 on Debian Etch i386 with the Digium's Dev Kit that comes with 1 FXO and 1 FXS. How do I configure my home PBX in such a way that whenever someone calls on my trunkline (PSTN) number, he/she will hear a customized ring tone, probably playing an MP3 file, instead of a boring standard ring tone while the extension number that is