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...3PSoundChannel6IsOpenEv Jan 3 17:06:26 WARNING[5817]: loader.c:440 load_modules: Loading module chan_oh323.so failed! I've tried to upgrade to version 0.6.5, but i got a compile error. Anyone know how to solve this error ? Thanks in advance, and have a GOOD 2005 Regardz, Jo&atilde;o Amaro </pre> </body> </html>

...isk-1.0.3 #make && make install && make samples Asterisk-oh323 #cd asterisk-oh323-0.6.5 Edit the Makefile #make && make install && ldconfig _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jo?o Amaro Sent: Thursday, January 06, 2005 7:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk - oh323 driver Hi all I've managed to get chan-oh323-0.6.5 working with asterisk-1.0.3 I've downloaded all the files from www.inaccessnetworks.com...

Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! Mark p.s. there was no 0.6.0 release.

hi I really want a good queue_log analyser, but I don't want to waste > EUR 1000 for something like that, so I thought starting a small project for it. I started off in php just creating a basic parser for the log, and I'll go on extending it. See http://karlsbakk.net/asterisk/scripts/queue_log_analyser-0.0.1- pre1.php.txt Does anyone want to join this effort? roy

I'm following the instruction from Jo?o Amaro from the page http://lists.digium.com/pipermail/asterisk-users/2005-February/090752.html Everything went fine until I run the 'make' command under asterisk-oh323-0.6.5. I got the error message chan_oh323.c:5220: too many arguments to function `ast_channel_register' I have attached...

...e call between SIP phones ? And with H323 phones (with a gatekeeper) ? And if i want to make a call from an H323 phone to a mobile phone (GSM) ? Are there any relation between codecs and protocols ? Which is more efficient ? If anyone have a FM, please tell me so i can RTFM. Best regards, Joao Amaro - Braga, Portugal ___________________________________________________________________ O SAPO j? est? livre de v?rus com a Panda Software, fique voc? tamb?m! Clique em: http://antivirus.sapo.pt

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I'm using the apllication AgentCallBackLogin so agents can login to a queue. They just need to enter the password, the CallBack Extensions is the ${CALLERIDNUM} Is there a way to AgentsLogOff withou using the AgentCallBackLogin application. I don't want the user to enter they CALLERIDNUM. Regards -----BEGIN PGP SIGNATURE-----

...port, i just want to receive/place calls to Cisco CallManager. If anyone tried to install OH323 with asterisk RC2 with success, please send-me an email. I can compile the driver and the library, but i can't initialize the driver when i start the asterisk. Thanks ind advance, Regards, Jo?o Amaro -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBODD/JUm/Bor63CERAsYrAJ9BUydM1fCRVDZIljpP7efvuARiLgCgp+LO UCuqUBRPCJMfyAtGZXPhb1c= =wXUK -----END PGP SIGNATURE-----

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list What's the syntax of the QueueAdd command for the manager ? I want to add interface FOO to queue BAR using a manager. Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBWDAVJUm/Bor63CERAhbtAJ0YykdT+TNTNKXlC3NldtzCWLo1owCfZEoz

...are broken.<br> <br> However i can transfer the call using the # key (via asterisk), but i<br> want to know if is possible to do this using the<br> cisco transfer key (via ccm).<br> <br> <br> Thanks in advance<br> <br> <br> Jo&atilde;o Amaro<br> <br> <br> <br> -----BEGIN PGP SIGNATURE-----<br> Version: GnuPG v1.2.4 (GNU/Linux)<br> <br> iD8DBQFB9j/OJUm/Bor63CERAnMYAJ9ww1VHxZ/YP8fIurUTMcFxrp8IoACfbvj/<br> VwA59Os8h5SLmr67YwMn1wI=<br> =0/WZ<br> -----END PGP SIGNATURE-----<br&g...

...dies, withou any error message, nothing at all. But, if i start asterisk with cli console (-vvvvvvc) theres is no problem and i can create the conference rooms. Zaptel and ztdummy modules are already loaded. Anyone with the same problem ? Asterisk: 1.0.8 Kernel: 2.6.9-5.0.3.ELsmp Regards, Jo?o Amaro ====> DUMP <==== [root@VoIP1 ~]# service asterisk restart [root@VoIP1 ~]# asterisk -vvvvvvvvvvvvr (...) ~ -- Executing MeetMe("OH323/R47", "1234|ciMps|") in new stack ~ == Parsing '/srv/etc/asterisk/meetme.conf': Found 2005-06-28 16:49:53 WARNING[4555]: chann...

Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the "pleeps" on the phone but asterisk doesn't intervene with the "Transfer" prompt. Am I missing something? Thx!

...to ogg format in the most efficient way? Right now I am doing mencoder dvd://1 -oac copy -ovc copy -alang es -slang es -o ~/DVD.avi ffmpeg2theora --optimize -S 0 -K 500 ~/DVD.avi Is there a way to avoid the avi step? (the alang and slang is to choose the Spanish version of the DVD) Thanks! Pau Amaro-Seoane

I have installed Asterisk@Home on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I just wanted to report that I organised a meeting last September and we recorded all talks and I asked computer support to convert everything into ogg theora... Maybe you want the link?? The talks are also very interesting if you are interested in Astrophysics and black holes... http://www.aei.mpg.de/~pau/LISA_Astro-GR@AEI Look under "programme" Long life to ogg! ;) Cheers,

I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I putting two * boxes into production. It's a callcenter + voicemail to Cisco callmanager. My problem is that mpeg123 sometimes doesn't terminate. What should i do ? Don't use MusincOnHold, and use a single MP3 file with a high length ? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with