On Tue, 2006-05-09 at 02:03 +0800, Kenige Ho wrote:
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
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
Subject: Re: [Asterisk-Users] Fake Ring Tone/Compile AddonTo: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Kenige Ho wrote: Dear All, I am currently have
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
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
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
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
HiAsterisk-Users,
Please help as it is out of my league and understand as why the call would be silent or partly silent (calling party can't hear the called party, but called party can hear). Thanks in advance.
Regards,
Kengie
On 1/19/06, Kenige Ho [EMAIL PROTECTED] wrote:
Dear All,
I am
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
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
,
Kengie
On 12/13/05, Kenige Ho [EMAIL PROTECTED] wrote:
Dear All,
I am trying to get a variable via AGI GET VARIABLE , but using AGI DEBUG I actually do see the variable get return but somehow my retrieving the variable via php. I don't get the value of the variable. Below is my code and my results
Dear All,
I am trying to get a variable via AGI GET VARIABLE , but using AGI DEBUG I actually do see the variable get return but somehow my retrieving the variable via php. I don't get the value of the variable. Below is my code and my results. Please help. thank you.
Coding:
on.
I want to thank the person that left that tip in the mailing list.
Sorry I forgot who it was as I was searching through the entire
archive from the begin.
I hope that this will help some people when there isn't any one to help you.
Regards,
Kengie Ho
On 7/21/05, Kenige Ho [EMAIL PROTECTED
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
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