similar to: why can't my asterisk restart?

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk. Oh323 Module compiled without errors. But When I try to stary Asterisk with the Oh323.so file in the modules folder, Asterisk is dying with the following error. [chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490 monmp3thread: Request to schedule in the past?!?! => (InAccess Networks OpenH323 Channel Driver) ==

Hello, I'm trying to register with a gatekeeper using chan_h323, I have a Login, Password and a user (telephone) number, This is my h323.conf [general] port=1720 bindaddr=x.x.x.x (my fixed IP) gatekeeper=x.x.x.x (gk ip) allowgkrouted=yes allow=all [xx] (my login) type=h323 e164=111 (my assigned phone number) secret=1111 (my password) context=incoming I'm getting this error when trying

Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikel&cat=270174&id=10&artlang=en I works my asterisk well before install the chan_oh323.so. But after I do "make install" the oh_323, my asterisk crash and show me the following message (asterisk -vvvvvvc). Does anyone have any idea about it? What's wrong

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI> load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI> How can I solve

Hello! With today's swfdec, Yahoo video is working! The upper half of the screen is still covered with error messages, but the bottom half is showing: http://video.yahoo.com/video/play?vid=448483 Google video shows some progress too. The slider is working now. Good job! Thank you! -- Regards, Pavel Roskin

Hi. I'm using Asterisk & oh323. My gatekeeper needs h323 ID and e164 ID for registration. But I don't know how to send e164 ID. I configure register section in my oh323.conf file like below. [general] <<snip>> gatekeeper=xxx.xxx.xxx.xxx [register] alias=abc alias=777 <<snip>> And the gatekeeper doesn't need passward. It authenticates to use only both h323

Hi. Download pwlib, openh323 Janus-patch at http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries Refer how to compile http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html Cheers. Kim. > > Message: 1 > Date: Wed, 11 May 2005 16:42:25 +0100 > From: Cesar Garcia <cesar.garcia@idecnet.com> > Subject: [Asterisk-Users] oh323 driver compiling

Hello. Though I pressed # after I finished recording my voice, but application record() did not stop. dialplan is : exten => 0,1,Record(recordyourmessage.gsm) exten => 0,2,Playback(recordyourmessage) exten => 0,3,Hangup I don't want either silence and maxduration time. Can you help? Regards. Kim.