search for: shilliday

...connects were caused by * bridging the calls and then the connection through the firewall timing out in some way. I tried setting reinvite=no on the SIP phones so that the audio would be routed through *, but it didn't seem to help. Does anyone have a suggestion for a next step? Thanks. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:...

Hi all, I've installed Asterisk on Linux Red Had 9. Now, I was trying to set up a GUI based system for the PBX. I downloaded some packages, but I have to have Perl running CGI scripts through the webserver. It does not allow me to. I am able to run a basic script that just just prints out html messages and nothing else. However, when I try to run am-main.pl or config.pl or any other

...in new stack -- Called Cisco403b9e -- SIP/Cisco403b9e-d3e0 is ringing == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/2-1' in macro 'stdexten' == Spawn extension (incoming, s, 4) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970

...n in zapata.conf. So, if my understanding is right, can someone please suggest a strategy for adjusting the gain controls? There are two controls, and each can be adjusted up or down. I'd like to adopt a method other than random fiddling. Where's a good place to start? Thanks! Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970

So Asterisk can't send VOIP calls to Vonage -- but it is still possible to use Vonage for flat-rate long distance by connecting the Vonage AT-196 to an * FXS port, right? The price is an extra D/A <--> A/D conversion. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Tim Petlock [mailto:tim@petlock.net] Sent: Monday, October 04, 2004 9:03 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk...

Anyone have some good examples if passing a call from one switch to another using IAX. I would like to have a call come in over PRI and pass it into a certain context of another server. Bother server seem to register with each other fine... I thought the command was... switch => IAX2/user:secret@host/context but I can't get it to work. Can anyone shed some light? Thanks, Chris

...dle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Justin Carlson [mailto:justin@lach.net] Sent: Tuesday, December 14, 2004 4:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uniden UIP200 Hello all,...

...s), renamed it to P0S3-07-1-00.bin. No, that's NOT a misprint. I renamed P003-07-1-00.bin to P0S3-07-.bin -- in Version 7.1, P003 isn't the Call Manager software, it's the Universal Application Loader that needs to load first to install everything else. 4. Rebooted the phones. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Daniel Poulsen [mailto:dpoulsen@gmail.com] Sent: Monday, September 20, 2004 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: mboehm@cytelcom.com...

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

Hello, I cannot accept any inboud calls from any provders in my asterisk which tries to authenticate the provider and at the end rejects the call with tthese message 407 Proxy Authentication Required How do I turn off this message. Thanks. Ehsanul Karim