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...he type file-gr.gsm. However, although the SetLanguage(gr) application is executed, only the plain file.gsm files are played. Note that I haven't setup gr as a language in any other configuration file. Any clues are welcome. Thanks Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: anpanag@hol.net

...es any one know how I can get the IP of an incoming channel in order to be able to dial back to that channel after the call is hangup? Does registering with a gatekeeper has anything to do with this? Any clues welcome Thanks, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: anpanag@hol.net

...5512, 5) exited non-zero on 'H323:5404' -- Hungup 'H323:17089' == Spawn extension (default, 4302, 1) exited non-zero on 'Local/4302@default-ee1a,2' -- Hungup 'H323:5404' Thanks in advance, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: anpanag@hol.net

Hello everybody, I am trying to setup asterisk to redirect international calls via a carrier which uses a fixed price tel number. The scenario is Dial 087..something (UK number) Pause for answer at the other end Send required telephone number 003..etc followed by # What is the easiest way of doing this? I have trouble with both the pause and adding the # at the end of the number. Best

I have been struggling with my Asterisk setup for 3 days now and I think I have done well...apart from the small detail that I cannot dial out on my phone (PSTN) line. My setup is: Suse Linux 9.0 1 fxo card connected to a BT(UK) line 1 Cisco ATA186 sip v3.0 with two analogue phones attached to it Asterix CVS-HEAD-05/30/04-06:56:31 with the UK Userid patch applied. Asterisk loads without any

I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20

Thanks for the reply Bill. I am aware of the interrupts problem. To solve it I have already disabled my serial ports freeing up interrupts 3 and 4 and these are allocated to the two cards. This was done 2 months ago and has not solved the problem. Is there any way that something can wake up every now and then and generate these two interrupts? My current /proc/interrupts is as follows: