To help eliminate issues related to libraries, etc, I moved back to a windows based machine and the windows based version, connected to the pbx with a basic gig-e switch. I did a full install instead of a server only install then copied the conf files over from my linux box. The Yate client rocks - the logging was extremely useful (and something the other freebie h323 clients didn't have). The Yate client worked fine going locally on the Windows Yate server and it worked fine from other systems on the LAN. I'm going to just keep it on a windows vm for the time being :)
I think a big part of my problem has to be NAT. There is a public and private network segment here, I was putting Yate on a server on the public segment and my pbx was on the private segment. The router config is quite odd and lots of two way NAT'ing is involved, even opening up all ports between the two boxes didn't do the trick.
Now comes the fun part, tying in an Avaya box from the Czech Republic :)
Thanks for all your help Paul, your suggestions were very useful and helped me work through this issue.
--
Rob Koliha
Development and Infrastructure Services
AVG Technologies USA, Inc.
-----Original Message-----
From: Rob Koliha [mailto:Rob.Koliha-***@public.gmane.org]
Sent: Thursday, July 29, 2010 3:36 PM
To: yate-uHKunLg9Q/***@public.gmane.org
Subject: RE: [yate] Basic questions, h323 to SIP
Sorry meant to cc to the list, this info/troubleshooting may come in handy for someone else :)
--
Rob Koliha
Development and Infrastructure Services
AVG Technologies USA, Inc.
-----Original Message-----
From: Rob Koliha
Sent: Thursday, July 29, 2010 3:34 PM
To: 'Paul Chitescu'
Subject: RE: [yate] Basic questions, h323 to SIP
Great suggestion - it doesn't appear to be listening, and it doesn't look like there is anything with h323 in the filename in /usr/local/lib/yate/*
[***@www ~]# netstat -nlp | grep yate
tcp 0 0 127.0.0.1:5038 0.0.0.0:* LISTEN 12330/yate
udp 0 0 0.0.0.0:5060 0.0.0.0:* 12330/yate
udp 0 0 0.0.0.0:4569 0.0.0.0:* 12330/yate
[***@www yate]# cd /usr/local/lib/yate/
[***@www yate]# ls -amlR | grep h323
[***@www yate]#
If I look in the folder I extracted the original distribution in I see "h323chan.cpp", but it doesn't look like it was compiled (no corresponding h323chan.yate file like the others). Others that are missing the .yate files are amrnbcodec.cpp, faxchan.cpp and gsmcodec.cpp. So I re-ran config. I know from reading through some of the make files that if pwlib/openh323 are not there it will do some funky stuff.
checking for Pwlib in /usr/local... installed 1.10.3 RTTI: none
checking for OpenH323 in /usr/local... no
So I ran: make distclean
Then I ran ./configure --with-openh323=/root/openh323_v1_18_0/
It found OpenH323, and now shows this when Yate starts:
Loaded module H.323 - based on OpenH323-1.18.0
Netstat now shows:
tcp 0 0 127.0.0.1:5038 0.0.0.0:* LISTEN 1993/yate
tcp 0 0 0.0.0.0:1720 0.0.0.0:* LISTEN 1993/yate
udp 0 0 0.0.0.0:5060 0.0.0.0:* 1993/yate
udp 0 0 0.0.0.0:4569 0.0.0.0:* 1993/yate
udp 0 0 0.0.0.0:2427 0.0.0.0:* 1993/yate
I'm still getting "The Gatekeeper Cleared the Call" from Ekiga when I attempt to dial, and nothing is logged relating to h323 in the log. I do see the <sip:INFO> entries if I attempt to make a SIP connection though. I added the h245tunneling=yes to see if that helped, ran into the same results. I also did a simple telnet test to port 1720, I can connect locally from the linux machine and from the test workstation I'm using Ekiga on.
There is no firewall enabled on the linux machine, and I received the same results (relating to h323) when using the pre-compiled windows version of Yate on a machine connected to the same switch the digium pbx is (to eliminate NAT or firewalls as a cause).
Any other ideas?
Thanks again!
--
Rob Koliha
Development and Infrastructure Services
AVG Technologies USA, Inc.
-----Original Message-----
From: Paul Chitescu [mailto:paulc-uHKunLg9Q/***@public.gmane.org]
Sent: Thursday, July 29, 2010 2:15 PM
To: yate-uHKunLg9Q/***@public.gmane.org
Cc: Rob Koliha
Subject: Re: [yate] Basic questions, h323 to SIP
Rob,
Do you see any H.323 related message in the logs? Do the H.323 calls from Yate
work (SIP->H.323)?
Try netstat -ntlp (as root) - does it show Yate listening on port 1720? You
said something about that but I'm not absolutely sure.
If Yate listens on that port you may have a firewall blocking the incoming TCP
connections. H.323 uses many connections (both TCP and UDP) in both directions
for each call so I suggest allowing all ports in both directions and
selectively blocking only the known services you want to protect.
You can reduce the number of connections by setting h245tunneling=yes in
section [ep] of h323chan.conf - of course if the other party supports it too.
Tunneling is also required if the connecting client is behind a NAT - although
that configuration is not really supported and you may run into voice trouble.
Paul
Post by Rob KolihaHi Paul,
Thanks very much for your help. I made the configuration changes you
suggested and after some more troubleshooting, I still seem to be running into
the same issues. I thought that it could be a connectivity/firewall/routing
issue, so I also attempted to get this going with the windows version of Yate
on a machine that is in the same subnet as the digium pbx (with the same
results). I figured that using the windows version at minimum would help to
eliminate the possibility of my versions of pwlib, openh323, or something else
relating to my linux server as the culprit(s) for this problem.
Post by Rob KolihaI have also tried removing the rules in the regexroute.conf file to allow me
to register to yate with an h323 client to do basic internal testing (self to
self or self or self to test numbers). That's not working either, same
symptoms with PacPhone -- GK Transport Error (or when dialing from Ekiga -
"Gatekeeper Cleared the Call"). I can register without issue via SIP. I'm a
novice at h323 and Yate is new to me, but based on the behavior it seems like
the establishing the h323 connection is the main issue - not the routing or
the SIP side. It's listening on 1720 at least, but I'm not seeing much more
:)
Post by Rob KolihaIf I attempt to make a SIP connection when running Yate as ./yate -vvvvv I
see on the console: <sip:INFO> Received 324 bytes SIP message from
ip.address:3344. If I attempt an h323 connection, the console doesn't display
any messages whatsoever. I have debug logging turned up to 9 for h323, is
there somewhere it dumps the logs to on the system?
Post by Rob Koliha<cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/2'
in 254 usec
Post by Rob Koliha<sip:NOTE> Formats for 'audio' changed to 'mulaw' [0xf1dcd0]
accfile.conf, h323chan.conf, regexroute.conf, regfile.conf, yate.conf, and
ysipchan.conf. There are empty lines between the [ ] sections, I've just not
included them with this message to make it easier on the eyes.