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...t is going 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 <kengiepanda@gmail.com> wrote: > 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...

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

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 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

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

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:

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

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