similar to: How to use same h323-conf-id in incoming and outgoing legs?

hello all, I would like to create an application where a file is played repeatedly in the background while two parties are having a conversation. Does anyone know of a way to achieve this? I have been looking into the ManagerAPI to redirect the call to a meetme room. Then I try to introduce a third party that will just play the file continuously, but with no luck. Is there a different/simpler

Hi, I'm trying to install asterisk with h323 support on rh9 box. I want to find a working combination between asterisk,asterisk-oh323,pwlib and openh323. Thank you! Ginel Tudorache

Hi all, After having tested the SIP part (successful :-)) we are now testing the H323 part of Asterisk. The H323 channel is up and running (using NuFone Network's Open H.323 Channel Driver)) However when dialing to * using ohphone the call can not be set up / established /H323 debug enabled *CLI> == New H.323 Connection created. -- Received SETUP message...

> > make sure you only use one driver e.g. either chan_h323/chan_oh323 > otherwise > > you'll end up crashing * Perhapas I haven't understood what you mean... I've downloaded the code on the inaccess Networks WEB, that is the chan_oh323, isn't it? I haven't installed it yet !! And so, where is the other driver?? Is it another code? In another WEB site? Thank

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

I m new to asterisk n i've got an IP phone that supports h323 protocol.... but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones.... but there is no h323.conf in /etc/asterisk/ .... i read that i've to compile some files but i m confused regarding h323 & oh323 ...... which one should i use.. plz tell me or atleast give some helpful

Hello, I am using Dovecot ver.1.0.7 on an x86 server with RedHat Linux Enterprise 5 and the following configuration: # 1.0.7: /etc/dovecot.conf protocols: pop3 login_dir: /var/run/dovecot/login login_executable: /usr/libexec/dovecot/pop3-login mail_location: mbox:~/mail:INBOX=/var/mail/%u mail_executable: /usr/libexec/dovecot/pop3 mail_plugin_dir: /usr/lib/dovecot/pop3

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

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does

I'm trying to compile channels/h323 and chan_h323 from CVS-STABLE, on Fedora Core 3. Firstly, despite the warnings in h323/README, I decided to try using the distro-specific versions of openh323 and pwlib. Of course, the Makefiles in channels and channels/h323 assume that openh323 and pwlib have been specially compiled in $HOME, so I modified the Makefiles to look for headers and libraries in

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Hello all, We have a Mandrake 10.1 server with Samba 3.0.13. The box has 2 NICs, 2 IPs and our DNS servers have both entries (same name for 2 IPs). Since kerberos is not working around here (tried to fix it for about 3 weeks, then gave up) I found my workaround: I use NetBIOS alias for my server, all clients use this alias for samba access and they authenticate thgrough SSPI. My problem is that

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

If I have a call coming in on one line and then I am using Dial to send it out to another outside line, is there a good way to get separate CDR for both call legs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060317/ac95b2fc/attachment.htm

Dears, We have a mail server (RHEL 6.0, Postfix and Dovecot 1.0.7). The output of dovecot -n is the following: # 1.0.7: /etc/dovecot.conf protocols: pop3 login_dir: /var/run/dovecot/login login_executable: /usr/libexec/dovecot/pop3-login mail_location: mbox:~/mail:INBOX=/var/mail/%u mail_executable: /usr/libexec/dovecot/pop3 mail_plugin_dir: /usr/lib/dovecot/pop3

Hello. I use rsync version 2.4.6, protocol version 24 on a RH7.2 machine. The purpose is mirroring a Red Hat repository. All works fine, *except* for really big files. ISO images (~650MB) are a good example. Whenever I don't exclude *.iso, I get the following: ================================================================== receiving file list ... done sending sums for 1408