Displaying 20 results from an estimated 10000 matches similar to: "FW: Asterisk@home - requesting help on oh323, ISDN BRI and iConnectHere DID"
2004 Jul 14
1
oh323 dial structure and oh323 debug?
According to the wiki at voip-info.org, the dial structure for using oh323
without a gatekeeper is:
OH323/<exten>@<host>:<port>
or
OH323/<exten>
The second option is valid only in the case where a gatekeeper is used.
NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination
host. When this version is used then the above syntax should be:
2003 May 23
3
iConnectHere - calls dropping out?
Hi all,
This is my first post here - I started with Asterisk a few days ago and have
"fallen in love" - fantastic product. I've only got softphones connected at
the moment - I'll probably order the FXO/FXS cards in about a month (and
then think about getting some hardware SIP phones). Our current phone system
is quite a few years old and isn't growing with us (when a single
2003 Jul 08
2
oh323 problem (small one)
I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.
When I try to dial an outbound oh323 call I get the following error :
-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's'
2004 Apr 28
2
Asterisk and Iconnecthere pause
Hi, I just got a SPA-2000 in and was finally able to complete my asterisk
setup. I'm making my outgoing calls through iconnecthere from the
asterisk server however I'm running into a problem when placing calls. I
can connect fine but when the person (or answering machine) picks up I
hear them talk for a about half a second then there is a half a second
pause or muted period and then the
2004 Jul 29
1
OH323 and codec selection
I'm having a small issue with the oh323 implementation when it comes to
codec selection.
Version info:
CVS Head 6/30/2004
OH323 0.6.3
OpenPhone for windows version 1.8.1
Asterisk is configured as a h323 endpoint which either terminates to the
PSTN locally through a PRI or terminates the h323 call to an IAX provider
remotely. Asterisk also has G729 licences installed.
in oh323.conf we
2004 Aug 05
1
iConnectHere and CallerId
Is it possible to send the CallerId to IconnectHere with Asterisk
when making outbound calls?
I read somewhere that it doesn't work.
I have set up everything to send the correct CallerId info to IconnectHere
but I get a "442-887-926267" caller id.
In [globals]
ICONNECT1=1713...(my number)
MYNAME=My Name
I set up the Caller Id in the dialing macro:
[macro-iconnecthere]
exten =>
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,
2005 Sep 21
1
oh323 driver and RFC2833
Hello,
I have installed oh323 channel driver. Outgoing calls to H.323 world do not
include RFC2833 in the outgoing TerminalCapabilitiesSet despite that
userInputMode=RFC2833 has already been set.
Does anyone know how to make RFC 2833 DTMF relay work over oh323 channel?
Kind regards,
Fernando Herrera
_____
De: Fernando Herrera [mailto:fherrera@iplan.com.ar]
Enviado el:
2004 Jun 02
1
oh323: Failed to create smoother
Hello,
I tried to get the oh323 drivers running. The driver loads, but as soon as a
H323 voice communication should be started, following error occurs:
-- Executing Playback("OH323/R1", "invalid") in new stack
Jun 3 01:26:20 ERROR[294931]: chan_oh323.c:1933 oh323_write: OH323/R1: Failed
to create smoother.
Jun 3 01:26:20 WARNING[294931]: file.c:539
2004 Jul 30
2
asterisk-oh323-0.6.3a
Hi there.
I thy to compile asterisk-oh323-0.6.3a but it fail in the make command.
I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README
file of the packet asterisk-oh323-0.6.3a
I do make and this is the error:
# make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target
2006 Apr 08
2
oh323.conf problem
I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2).
When I put codec (*3) Asterisk doesn't start (*4).
What have I done wrong? I
2003 Sep 05
1
oh323 call segmentation fault
hello,
i have problem with oh323 channel driver (tryied differnet versions).
when i try to make call between oh323 - sip, oh323-isdn, oh323-capi
asterisk crash with segmentation fault. Channel driver was compiled with
pwlib 1.5.0 and openh323 1.12.0 libs.
Does anybody know solution ?
WrapH323Connection::WrapH323Connection: WrapH323Connection created.
-- Executing Dial("H323:31119",
2004 Nov 25
2
oh323 compile issue
Hi all,
I want to give a try to oh323 (currently nufone h323 channel is setup
and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib
1.8.1_ and _openh323 1.15.1_ What I made:
openh323 dir:
make clean
apply the oh323 patch
configure
make opt
asterisk-oh323-0.7 dir:
make
[...]
wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel
(H323Connection &, int,
2004 Sep 14
3
OH323 Trunking
I've successfully got inbound/outbound calling working with our Asterisk
using the Asterisk-OH323 channel driver. We are using a parent gatekeeper
and the NuFone H323 channel driver would not work with the parent
gatekeeper...
I'm trying to determine a way to ensure that the line used for outbound
calling is always available i.e. like trunking..
>From what I can tell when I place an
2004 May 06
4
asterisk-oh323, new version 0.6.1
Hello all,
This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio"
problem of the previous release.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
2004 May 31
1
Failover: iconnecthere to voicepulse
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect
exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN}
Well,
2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
otherparty realtime voice , but other party geting sip party's voice 1 sec
later (not
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
2003 Nov 07
3
Unable dial out with the new Oh323 0.5.6
Hi all,
i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then
i've installed the new chan_oh323 (0.5.6).
when i try to make a call with "netmeeting" through * ( * dial out with
"Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked.
Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7)
installed, and it worked.
Is here
2003 Jun 08
1
oh323 and extentions.conf
>
> hi
> i am not using sio or iax but only oh323. i am trying to register my
> extensions like
>
> extensions.conf
> ;-- H.323 [alias = 665]
> exten => 665,1,Dial(OH323/172.18.1.133)
>
> oh323.conf
>
> context=voip-h323
>
> ;-----------------------------------------
> ; Configure H.323 aliases, prefixes and
> ; related ASTERISK's contexts