similar to: Pickup() h323

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more,

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! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", "SIP/6166 at asterisk") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/6166 at asterisk > 0x7fa9d4007660 --

Does asterisk work with CCM as gateway ? When I trying call asterisk,I totally can't hear any sound. When call ohphone - works good. 10.0.1.219 is CCM, 10.0.1.207 asterisk. Trace messages here : -------------------- == New H.323 Connection created. -- Received SETUP message... == Setting up Call -- Calling party name: [5001,] -- Calling party number: [5001] -- Called party

hi all im trying to compile h323 i have got the pwlib and openh323 working that is simph323 is running properly but when i try to compile h323 in the channels directory it gives me the following error can anybody please help me with [root@test src]# cd /usr/src/asterisk/channels/h323/ [root@test h323]# make opt g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_LINUX=2.6.5-1.358

Just a warning to people trying to use ooh323 with Asterisk 1.8.7. The example config file that comes with asterisk is called chan_ooh323.conf when it actually should be named ooh323.conf for it to work. Sent me into a panic when I was trying to install an H323 link to an Avaya server and the ooh323 module would not load because it could not find its configuration file. The file needs to be

05.03.2015 11:42, Dmitry Melekhov ?????: > 05.03.2015 11:29, Dmitry Melekhov ?????: >> Hello! >> >> Just installed asterisk 13.2.0 and see many such messages in log, I >> see them in console during calls, really something like this: >> >> >> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", >> "SIP/6166 at

hello every body, i have big problem to configure h323 trunk between cisco router and asterisk 11.13.1 which uses ooh323 module. any body knows if ooh323 module can work with cisco routers or not???? (in gateway mode, it is ok and register in cisco gatekeeper but i can not configure trunk h323) any comments or hints are really appreciated. SAM -------------- next part -------------- An HTML

I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN). When chan_ooh323 first loads it tries to establish a connection with the gk but I it fails. I have the following extract from the ooh323 log. Can anyone give some insight? Thanks! MD 23:02:59:045 Sent GRQ message 23:02:59:045 GkClient Received RAS Message 23:02:59:045 Received RAS Message = { 23:02:59:045

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

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

Has anyone here any experience in getting an Asterisk box to talk to a Cisco Unity system? I have a potential customer who would like to add a conference bridge to their existing Cisco Unity setup. The digging I have done so far suggests that it should be possible to talk SIP between them, but I'd be interested in any stories of success or failure. Cheers Tony -- Tony Mountifield Work: tony

Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide

I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read in another thread I tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package). This doesn't seem to work to

Has anybody successfully (or I guess unsuccessfully for that matter) implemented Cisco Call Manager and used an * box for voicemail? I checked the wiki and google and I see some references to Call Manager Express and *, but CME is completely different than CM. If anybody has done this or has any insight, it would be appeciated. We are trying to migrate ~ 300 users off of Cisco Unity and

Hi, I'm trying to connect Asterisk 1.4 and Cisco CallManager 5 using SIP Trunk without MTP (media termination point). Howerver, Cisco 79xx phones do not support RFC2833, they always notify CCM5 via SKINNY channel no matter where they send RTP to. For non-MTP trunk there's Out-of-band DTMF support in CCM5 called "kpml". I wonder if Asterisk can support it. I found an

Hello! I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I tried to do "make" I got such error: * chan_ooh323.c: In function `reload_config': chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_ooh323.c:2053: error: (Each undeclared identifier is reported only once chan_ooh323.c:2053: error: for each function it appears

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes