similar to: Latest version of asterisk

I tried to use it to create a 'trunk' to Cisco's call manager. The 0.7.1 code worked up to a point. The call would be established, but audio was one-way from the Call Manager. Asterisk with Chan_h323 would not setup the sending rtp stream. The debug results showed the sending stream as using ip:0.0.0.0 I have not checked for a CVS update to see if it is fixed, or if that

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -----Original Message-----

I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel. Did you install the kernel-sources and kernel-util rpms as well? You'll need these in order to compile and install zaptel. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of T. Chan Sent: Friday, January 30, 2004 4:01 PM

Hi, I have 2 internet connections (1 adsl/1 cable). I am try to route all outgoing mail from the mail server (on the same box), through the ADSL connection routing through the cable will mean mail will get rejected by AOL :( I am using qmail as the mail server. The configuration is: eth0 : cable connection ppp0 : adsl connection eth2 : internal lan connection I have configured split access as

the problem below keeps recarrying even after i have cleared this error when i run asterisk -vvv or -c the error occurs again please help ..Warning, flexible rate not heavily tested! .................WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File

Hello I am trying to configure my extensions.conf and oh323.conf to termination calls to a H323 service provider. Can anyone send me a sample config files? or tell me where to put the username and password which my service provider has given? also how to put the Dial command in extensions.conf Asterisk rocks. I have a SIP provider configured for all my international calls and it works absolutely

Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip<->h323 gateway to tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Hello, I have gotten an application to run well in Wine. However, when this application tries to print a text document, the size of the text is so small that it looks like dots on the page. It is not readable. The printing of graphics is better. Any advice on how to get the output text to be larger would be greatly appreciated. Aram J. Agajanian

Has anyone tried to wrap your SMB session inside another TCP protocol like SSH? I'm trying to sync files across two linux machines but my broadband provider is blocking SMB and port 139 traffic. If anyone has any information it would be greatly appreciated. Aram -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in

Hello all, i'm experiencing a list of unpredictables hangup on SIP phones using a PRI E100P Card. All i can see in logs is " WARNING[81931]: File chan_zap.c, Line 5402 (zt_pri_error): PRI: Read on 37 failed: Unknown error 500" I receive a lot of these errors in asterisk/messages. It doesn't seem to be strictly linked to hangups, since i have dozen of these messages per

Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA;

Dear all, I have had an experience which I would run by all of you to see if this is normal. I am running a few asterisk servers with 512M RAM memory, and as I have mentioned in previous notes, I have experienced frequent crashes when faced with more than 15-20 simultaneous calls. I have tried to find out if it could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3, (c) old

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

The Virus controllers wouldn't let me send the attachment... ----- Original Message ----- From: "Trevor Fraser" <trevor@systematic.co.za> To: "Tommy Eriksson" <tommy.eriksson@ctakt.com> Cc: <samba@lists.samba.org> Sent: Tuesday, September 10, 2002 6:32 PM Subject: Re: [Samba] Problem joing a W2KSP1 client to a Samba PDC (2.2.5). > Hi Tommy. > >

I converted the 128kbps tom.mp3 file into a .wav and then used oggenc -m 2 to laterally encode it to 128kbps tommy.ogg format. I noticed that it gets very close to the size of an mp3, great job guys. Then I got to thinking that maybe it could be compressed further with gzip so I tried it. I also gzipped the mp3 to compare and it looks like the mp3 gzips better but gzipping the ogg file will get

Greetings. I have problem with password expiration problem i cannot handle myself, so i wrote in this list. Recently i discovered that a newly created samba account has already expired password. smbldap-useradd -a -d /home/tommy -G education -s /bin/bash -M tommy -c "Tommy T." tommy smbldap-passwd tommy getent shadow user:*:::::::0 user2:*:::::::0 user3:*:::365::::0

Hi folks, I''m having trouble getting gems to install rails. It just keeps asking me which version of thin I want to install. Until I choose skip or cancel, in which case I''m told that something has gone wrong. Any advice? Thanks kindly, Tommy PS: Gems is 0.94. --------------------------------------------------------------------- tommy@tommy-laptop:~/www/blog$ sudo gem