Displaying 20 results from an estimated 8000 matches similar to: "Getting in to h323"
2003 Nov 28
0
Re: Resend: Help for oh323
Michael,
Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is
the latest :). Ok I will upgrade it. just for the record, following worked.
exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206)
Cheers
Sathya
Date: Fri, 28 Nov 2003 11:28:59 +0200
From: Michael Manousos <manousos@inaccessnetworks.com>
Organization: inAccess Networks
To:
2004 Aug 10
0
iconnect inbound - FIXED (kinda)
This appears to have been the magic bullet for me.
Thank you very much.
So, the bottom line is that there is a bug that ends up making inbound
calls use type=peer rather than type=user.
Correct?
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng
> Sent: Tuesday, August 10, 2004 8:35
2003 Nov 27
6
Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten => _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2004 Sep 16
0
H323 - Control Protocol Error (Master slave Determination)
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for using our H323 Endpoints which are ip200 Innovaphones.
Besides, we also use Gnomemeeting but
2004 Sep 07
3
H323 Control Protocol Error
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for using our H323 Endpoints which are ip200 Innovaphones.
Besides, we also use Gnomemeeting but
2005 Jan 05
0
One way audio [Asterisk + Innovaphone IP3000 + asterisk-oh323/h323]
Hello everybody,
I?ve been trying to solve a problem for several weeks now but it really
beats me.
There are several hard phones connected to an Innovaphone 3000 VoIP gateway.
On the other side I have a SIP softphone connected to Asterisk. The problem
I have is that on incoming calls (hardphones to softphone) I only have
outgoing audio (from soft to hardphone); everything is OK when I call the
2004 Jan 26
0
Anyone run * on OS X ?
With the 1U Apple G5 server at a good reliability/storage/pricepoint
level... got to thinking about compiling Asterisk on OS X.. at least for SIP
phone call switching, voicemail, etc. Has anybody attempted this? Email me
off list if this is too dev-heavy for the user list.
Thanks,
Ted W
-----Original Message-----
From: asterisk-users-request@lists.digium.com
2005 May 18
0
Asterisk and H323 vs OH323???
What is the difference between H323 and OH323 in Asterisk? I need Asterisk
to have basic H.323 support so we can offer some simple H323 termination
for some of our Cisco and Quintim hardware. Our upstream provider uses
SIP, so I figured I'd use Asterisk as the go-between. I already setup
Asterisk so it can push calls out through our providers via SIP. I just
need a good/solid/very simple H323
2003 Oct 27
1
SIP -> H323 Seg fault.
A very strange problem..
* dies with seg fault when calling from SIP to h323
WARNING[589848]: File chan_oh323.c, Line 2429 (alerted_h323_connection): Call with reference 183 in unexpected state (4).
-- Called 113506
-- H323:183 answered SIP/alex-1e48
Segmentation fault
I'm using Mandrake 9.1 , lastest * ( 3 day old from cvs) , and oh323 last version..
here are my configs:
2004 Sep 03
0
RC2 with OH323 or H323
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi All,
I've just finished my upgrade to asterisk RC2.
I need to have H323 support, and in the last months i've been using
the chan-oh323 with good results.
My question is: anyone in the list have made tests with both chans
(oh323 and h323), which is best ?
For this installation i don't need the gatekeeper support, i just want
to
2004 Aug 13
0
SIP<->H323 "Failed to create smoother"
hello,
Im tryin to make Calls from MS Netmeeting(h323) to
Xlite(SIP) it rings, but as soon as i answered it
dissconnects!!!!
This is what i get from the Asterisk console:
-- Executing Dial("OH323/R27469", "SIP/xlite1|10") in
new stack
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1265
create_addr: Setting NAT on RTP to 0
Aug 13 10:19:03 DEBUG[524304]: chan_sip.c:1500
sip_call:
2003 Dec 12
1
simple question on sip.conf
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another SIP proxy can
redirect his calls to my * as well. Those calls two will come through
general
2003 Sep 12
3
h323 v oh323
Use oh323.
Download the openh323 and pwlib tarballs from openh323.org
Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY!
good luck
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
sean.langley@gdcanada.com
> -----Original Message-----
> From: Senad Jordanovic [mailto:senad@cwcom.net]
> Sent: Friday, September 12,
2003 Sep 16
1
h323 gatekeeper registration failed
Hi all,
i have tried to connect to a clarent gatekeeper.
I have used both of h323 drivers chan_h323.so and chan_oh323.so.
But no one can register to this gatekeeper.
Our ip is activated on this gatekeeper.
Maybe, i do wrong anything....
I have only set the "gatekeeper" option in the h323.conf or oh323.conf to
the ip address from the gatekeeper.
gatekeeper=x.x.x.x
But no one of the
2004 Apr 13
1
SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
--
2005 Jul 01
2
How to Configure a H323 Phone (newbie here)
i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf
2003 Nov 24
3
Cisco to asterisk termination with h323 and g729 finally works.
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I tested both oh323 from inaccessnetwork and JerJers chan_h323.
I used 1.12.2 version of oh323 and 1.5.2 version of pwlib.
After latest changes from JerJer chan_h323.c works ok when receiving traffic
from ciscos. I havnt found any audio problems although I didnt send much
traffic.
Latest oh323 has some
2004 Jun 29
1
Registration of H323 Endpoints?
Hi,
I am using the asterisk-oh323 wrapper and I am looking to allow
registration of h323 endpoints and allow Asterisk to act as a gateway. The
idea is simple: H323 endpoints would register with Asterisk. They each would
have their own internal extension (like SIP). If a H323 endpoint dials an
outbound extension, then the h323 call gets routed to a H323 Gatekeeper which
then terminates
2004 Jul 22
1
Sip -> H323 using oh323 and G729
Hi All,
I have set up a box that will be used as follows:
SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server
192.168.1.5 192.168.1.50 192.168.1.80
Asterisk is running the latest CVS and oh323 driver.
The SIP phone is a Grandstream Budgetone 100.
I have everything setup and running with G.711 ALAW and ULAW and i'm able
to make calls through