Displaying 20 results from an estimated 500 matches similar to: "H323 dropping connections"
2005 Jan 06
0
H.323 to SIP extension
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 phone
to a SIP extension.
Here's the layout:
2005 Jan 27
0
Problem with OpenPhone->Asterisk
Hello all,
I just installed Asterisk with H323 support (chan_h323 from Jeremy
McNamara). But experience problem while connecting OpenPhone to Asterisk
Here is h.323 trace:
5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP
Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720,
handle=27
5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225
2005 Mar 03
0
I have met a message : "No one is available to answer at this time".
Hello, Users.
I loaded module chan_h323.so, chan_vpb.so.
I have met a message : "No one is available to answer at this time".
I don?t know what I do..
My 'h.323 trace 5' result is :
== vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
-- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack
1:21:34.936 ThreadID=0x06f2bbb0
2005 Jul 05
0
chan_h323 passes no audio?
I'm attempting to get chan_h323 working on Asterisk CVS-STABLE.
I've compiled it ok using the Janus release of pwlib/openh323, by
editing the makefile as per the comments.
Call setup and cleardown seems to work fine, but no audio is being
passed in either direction.
Doing an "h.323 trace 9", I noticed the following sequence at the end
of the call setup:
h323.cxx(1685)
2003 Sep 12
2
problem with * and Howlink CL-100 ip phone
I'm trying to use a Howlink CL-100 ip phone with *
It's h323 phone with very limited protocol support. But it's enough that I
can use it to dial netmeeting client and artisoft pbx just fine.
When I try to dial my * with it using either chan_h323 or oh323, it seems
to fail on negotiating H245. Maybe this phone doesn't support it?
I've used all different versions of
2004 Jul 06
1
* and Innovaphone
Hello,
I think I have the same problem as Martin Bene mentioned in
http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html
Since I found no further information about this I'd like to ask wether
you know what the reason for this problem is and how one can get around
this.
* is registered to the innovaphone gatekeeper.
Trunk connection is done with an AVM-B1 and chan_capi.
2004 May 18
0
problems with asterisk-oh323
Hello,
I've been trying to send traffic to a Cisco Call Manager 3.2, but with
no luck.
Here's whats happening:
* Call gets to CCM
* Call gets to the gateway
* Rings a couple times on destiny
* Call gets hungup.
On the CCM I get the following error: MediaManager - ERROR
wait_AuConnectErrorInd
On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not
available)
On asterisk:
2003 Jul 21
4
anyone with X100P & Callerid working outside US ?
I'm just curious if anyone has the X100P & Callerid receiving working
outside US.
Replies are appreciated. Also if it's not working for you in a certain
coutry you can respond too.
regards
Martin
2003 Jul 10
2
OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:729@216.52.153.206)
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to
2005 May 23
1
OH323 CONTROL PROTOCOL ERROR
>Please I have combed the Archive to no avail on this problem protocol
>control problem in oh323.
>I'm receiving calls from CISCO AS5300 -> Asterisk -> Zap Channel. The
>calls clears the remote location but drops on my own end. Please what
>could be
>wrong. I have included the oh323.conf and log files. I have tried
>various configuration and I thought I should
2003 Aug 27
0
Chan_h323/g729 - X100P connecting to non-Digium Partner
I have on Chan_h323 with G729 and X100P trying to connect to
a Planet VOIP400 gateway box(http://www.planet.com.tw)
I uncommented g729 in the Makefile and I'm setting g729 in h323.conf
I'm receving in my side:
1:20.906 H225 Caller:810f070 h323ep.cxx(1537)
H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser
and the other side(Planet) says:
15- RADH 2
2004 Aug 04
5
H323 Call Dropping
Hello All,
I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the
configuration:
CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK
My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk,
and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however
the gatekeeper drops the call
2003 May 31
1
oh323 problems
i am trying to make calls between two workstations using netmeeting and
asterisk.
i get the popup on both when i call the extensions 665 and 667 but when
accept, i get this error
*CLI> 0:18.190 H225 Caller:8112978 H225 Received connect
PDU.
0:18.288 H245:810b388 H245 Read error: Bad file
descriptor
0:18.318 H323 Cleaner H323
2005 Sep 09
0
woomera doesn't work (same OpenH323 problem as with chan_h323)
Banging my head against a brick wall trying to get a working H.323
implementation for CVS-HEAD. (The ONLY H.323 I have had working is
OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile
problems on OH323 for HEAD)
So, I thought, lets try this wonderful chan_woomera (dubbed "H.323
for Asterisk that works!").
I get exactly the same kind of problem as I have previously had
2004 Aug 11
7
H323 call dropped when answered
Hi All.
I'm using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) ---------------> Asterisk
-------------> H323 GK --------------> PSTN
I have tried all codec's and always the same result, the called phone
will ring without dropping for how ever I allow it to but as soon as it
is answered it immediately gets disconnected.
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
2005 Mar 16
0
Help with simple H323 settings
Hi,
I have about one year of experience with Asterisk, working with ZAP
(digium, junghanns) ZAPHFC, SIP and IAX. These technologies are quite
clear to me, the problem is that I have no experience with H323, but
now, I need to use this also.
The problem that I have is very trivial, so I think that this should
be a very easy question for you guys whom know how it works.
All I want to do,
2004 Jun 25
4
Failure in RTP streaming
hi,
I use the oh323 driver to answer H323 calls.
The connection is set up normally.
In my extensions.conf file I use:
exten => s,1,Answer
exten => s,2,Playback(demo-instruct)
exten => s,3,Hangup
So that when a call is answered i get:
*CLI> -- Executing Answer("H323/ip$10.0.3.23:32782/6502", "") in new
stack
-- Executing
2006 Oct 23
0
Callmanager 3.3(5) and Asterisk with ooh323 problem
I have searched and searched for over a week on this but can't seem to
find the issue. Calls from CallManager to Asterisk are being
disconnected immediately. I have setup CallManager and Asterisk per
Shaun Ewing's pdf
http://asterisk.edropbox.net/ccmasteriskvm.pdf
I have installed Asterisk 1.4.0-beta3 on Fedora Core 5. I got libpri,
zaptel, and asterisk compiled and installed.
2003 Jun 15
2
Voicemail with H.323?
Trying to configure voicemail with H.323 all I get is the following errors
when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it
might be a simple fix.
[chan_oh323.so] => (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/oh323.conf': Found
0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version
0.0alpha0 by inAccess Networks