similar to: How to test H.323

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

I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax machine. Both are connected to a DAHDI board. I'd like to route incoming PSTN fax calls to the extension of the fax machine and process non-fax calls through different dialplan.logic. What's the best way to go about doing this? I've looked into Fax for Asterisk, bit I'm not sure that I want it or NVFax

I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop firewall on a 5Mbps down/512 up cable connection. I'm having sound quality problems when users call in for voicemail and with music on hold. The sound is choppy and muffled while souding pretty good for calls inside the network. I'd appreciate some pointers as to where to start looking to improve things. I've

Hello, I have asterisk 1.4 with ooh323c addons installed. (As I am a newbie in voip world...my question might be idiot...! ;) Please forgive me!) I succeed to make H323 call when ooh323c is configured as gateway (gatekeeper=DISABLE in ooh323.conf). When I put gatekeeper= ip_address, and add an account as follow : [aaa] type=friend username=aaa password=xxxx host=dynamic context=test

I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've found the driver source code on https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is this code considered "somewhat ready for prime time use"? Thanks, Frank

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

Hi all, i have big problems to make a h323 call over the gatekeeper from my provider. The provider demanded following account data: H323 ID: XXX-XXX-XX-X DetinationNumer: XXXXXXXXXXX I have configured the oh323.conf following: gatekeeper=XX.XX.XXX.XXX alias=XXX-XXX-XX-X Isx the alias equal to the h323id ? And how i have to make a call with the dial app ? I have following config: exten

Hi, I'm using Asterisk-1.4.7.1. Everything was working fine. Now I'm trying to Install Asterisk-addons-1.4.2. The procedure I followed is as... # cd asterisk-addons-1.4.2 #./configure #make menuselect #make #make install Everything is going fine except make install. I've tried many times, but the same error I'm gettiing--- The error is--- asterisk-addons-1.4.2]# make install

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

I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It registers fine and I can call between the MP-114 and other extensions, but I'm not having much luck with the FXO ports. syslog shows the problem to be in the MP-114 configuration. Can anyone help?

I'm having a problem with my IAXy not always connecting to my Asterisk box. When I pick-up the phone plugged in to the IAXy I get a busy signal. I have to hang-up the phone and wait a few seconds after the orange LED goes out and then try again. When this happens I don't see any connection attempts in the Asterisk -r output. When I do get the IAXy to connect to Asterisk I get a

Hi, I have two asterisk servers where softphone A is connected to asterisk A. On those two asterisk servers, ooh323c is installed. I tried to call a "test" context on asterisk B from softphone A. But I always fall into context "default" of asterisk B. ( I don't know how to tell asterisk A extensions.conf to call asterisk B "test" context) Here are conf

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

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, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - AVS@210.21.118.XXX Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn-xxxx e.164 - 22xx2912

I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86de in ast_dynamic_str_thread_build_va (buf=0x8172763, max_len=0, ts=0x81482a0, append=0,

Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten => 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming calls to asterisk. However I'm not sure how to route calls to the remote h.323

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