similar to: asterisk 1.4 and gnugk with ooh323

Greetings to All, I hope someone has already gotten this working. I spent all day today trying to get ooh323 and gnugk to run on the same box. After a lot of tweaking to get everything compiled, I got both up and running. I can make calls IAX to H323, but cannot make calls in the reverse direction. I have tried many different configs on the GK, but always come up with the same error. It appears

I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER ---------------- All information provided in this

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Hi List, I have a client that wants me to replace their existing H323 gateway. I am able to get ooh323 and h323 to work fine in a native environment, but the whole thing goes to heck when I have to cross networks. Gnugk seems to be the answer to this, but I can't seem to get it to work right. Any ideas? Thanks Danny Nicholas

Can someone tell me how I can configure ooh323.conf to accept call from h323 gateway (only the authorized h323 gateway) to my asterisk. I will be glad to know how this can be done. I tried the setting as in ooh323.conf [abcd] type=user context=default ip=62.193.1XX.2XX disallow=all allow=gsm allow=ulaw this gateway can make call, even if these lines are commented out and you restart the

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

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Hi, I have problems compiling asterisk 1.4.11 with res_snmp. I do 'make menuselect', and I see that this resource module depends on netsnmp. I am using centOS 4.5. I do: > yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs I don't know if i am missing something. I go to the source directory and I do: ./configure but still does not work: > ... > checking for

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

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

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

Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing

hello Dmitry thank you for your reply. Ok, you are right. i want to configure trunk h323 between asterisk 11.13.1 and 2800 cisco router. this is my scenario: PBX(100)--->cisco--->asterisk11.13.1---->PBX(200) when i call from 100 to 200, everything is ok but when i call from 200 to 100, phone rings but after i answer it, i have no voice and call terminates after 5 seconds. this is

I'm trying to add ooh323c to my asterisk 1.2.7.1 install and did an svn update of asterisk-addons and followed the readme in asterisk-ooh323c and I get through the .configure with no errors. But make causes: rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread make: rpath: Command not found make: [libchan_h323.la] Error 127 (ignored) I'm not real sure what to try to fix

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

hi sorry about the urgent but it is urgent i have problems configuring a connection between asterisk and avaya using H323. the module i am usign is ooh323 what do you need to help me? and any tip or hint? thanks!!! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An

Hello all, I have problem. I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. OS:redhat EL4 Linux 2.6.9-5.EL #1 Wed Jan 5 19:22:18 EST 2005 i686 i686 i386 GNU/Linux Please help me . [root@asterisk asterisk-ooh323c]# make make all-am make[1]: Entering directory `/usr/local/src/asterisk-addons/asterisk-ooh323c' source='src/chan_h323.c'

Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as "plan B"). As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a