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Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the *

...uite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP packets. Regards, Jamie Carl Email: me@jazz-inc.net PH: +61-414-365-466 -----Original Message----- From: Tjardick van der Kraan [mailto:tjardick@vanderkraan.net] Sent: Monday, 19 May 2003 9:00 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Call between G.711 and GSM *This message was transferred with a trial version of CommuniGate(tm) Pro* The GSM codec in X-Lite is not compatible with the GSM codec...

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

The parameters 'force create mode' and 'force directory mode' have no effect when the option 'unix extensions' is active. The documentation does not report any interaction between the unix extensions and the forced modes, so I'm assuming it's a bug.I'd like to know if it is a known bug ?Has it been solved, if so in which version ? My server runs Debian lenny,

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

...t; > Today's Topics: > > 1. Re: Caller ID Problems (WipeOut .) > 2. Re: IAX, IAX2 and authenticatyion (Dan) > 3. RE: 7206 as SIP->PSTN Gateway? (Abdul Hakeem) > 4. Re: IAX, IAX2 and authenticatyion (Brancaleoni Matteo) > 5. Re: Dect Phone (Tjardick van der Kraan) > 6. Monitoring an active channel (Timothy Soos) > 7. Re: asterisk and defunct perl procs (Rich Adamson) > 8. Re: Caller ID Problems (Rich Adamson) > 9. UK Suppliers (Angel Gabriel) > 10. RE: UK Suppliers (Lee Redmayne) > 11. How to test * ? (Angel Gabriel) >...

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client