search for: redenlac

Hi Anybody have a example or use this modem as FXO, acording to intel this modem is a real full duplex modem, so i have one question why i can we use this modem as fxo My idea is simple when you recive a call in the modem you can transfer this call to any other place using for example a IAX2 or Openh323 protocol This is posible? Thanks

Hi When i make a call using oh323 is posible to make the ringing sound, to indicate the progress of the call At this time when i make a call, i can't hear anything unless time out or the call is made Please hellp!! thanks Sorry for my english

Hi What command i need to use to make a call with oh323 and hear the ringback sound Thanks

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should "talk" to Asterisk of course.. Thanks very much for some advices :) Bartek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running

When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030623/95bfcc74/attachment.htm

Hi I have a gw linux in this machine i have one quicknet card, how i can reserver a prt of my bandwidth to voice data, for example when i download a big file the voice don't loss quality thanks

Hi This bug is really crazy, please help me In the follow scenary ATA-186 -> SIP -> Asterisk -> SIP -> ATA 186 : No DTMF gets through * in outbound mode, Sip conf [204] type=friend username=204 secret=somesecretpassword host=dynamic canreinvite=no ; The follow line don't work dtmfmode=rfc2833 nat=1

Hi I buy this card just few month ago, but i have a problem when i try to use in the asterisk pbx the software detect when i plug or unplug the phone cord, but when i plug the phone cord just 2 seconds later asterisk detect a ring event but nobody is calling what happend??? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm