search for: tjardick

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 *

...'t that add quite 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...

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

Hi all, How do I route calls based on the DID the incoming caller dials? I?d like someone calling a DID to by-pass the main menu prompt and dial the extension associated with that DID directly. Thanks Michael Rose, Jr. ? ? ?

...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 something before speex works ? Greetings, Tjardick

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

Is there a way i can change my subcription email address, without unsbubbing and resubbing myself? Cheers, Mathew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030221/c9149135/attachment.html>

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

......" > > > 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 G...

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

Hi guys,

I have this configuration: UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet --> UA2 UA has provate address (192.168.x.x) Asterisk has public address I want to be reach somebody at the internet. My idea was that asterisk works as a Proxy. Then i would have a SIP/RTP connection between UA1 and Asterisk and an other SIP/RTP connection between Asterisk and UA2. (asterisk is

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

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

After battling for days trying to figure out what was wrong with my iax.conf it was determined that I do not have any inkeys set on the digium server. Now whether that is something new or just in a few cases I am not sure. Messing around and reading on IRC and the mailing list I could get certain things to work and then break other things. Now I can dial a IAXTEL number, 800 number and FWD

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back