Displaying 20 results from an estimated 400 matches similar to: "H323 no sound"
2005 May 24
0
H323 integrated Asterisk support
Hi all,
I used oh323 support from inaccess. It work very well.
I would like to test h323 integrated support.
This my problem when I test it :
I cannot heard any thing in both way.
The test is : SIP --> Asterisk --> H323
This is th debug trace from h.323 :
-- Executing Dial("SIP/someaccount", "H323/0033172897104@somehost") in
new stack
2008 Oct 18
1
strange h323 delay issue
Hello,
I have a strange h323 issue. After executing command
"Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18
22:32:23. Meanwile I have sniffing traffic on port 1720. The call was
established just at Oct 18 22:33:03 (New H.323 Connection created.) and also
packet sniffer grabs the h323 invites at this time also. So my question is
what
2005 May 16
1
Always Ringing
Hi all,
I am using chan_h323 from Asterisk CVS to interconnect with GNUGK
v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on
Asterisk. However, I only heard ringing when the call was answered on
SIP side. Below is the debug from chan_h323. Any help is welcome.
Thanks.
*CLI> == New H.323 Connection created.
-- Setting up Call
-- Call token:
2005 May 30
0
IAX2 to H323
Hi all,
I'm using following software and equipment and I have very strange behavior:
Asterisk CVS-NHEAD-05/30/05-16:42:41
H323 gatekeeper - GnuGK 2.2.2
IAX2 client - Firefly 1.9.8 build 3945
H323 client - SJPhone Build 1.50.271d
H323 gateway - Welltech Wellgate 3504A
When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected.
When I dial from SJPhone (H323) ->
2006 Nov 23
1
asterisk 1.4 chan_h323, help please...
Hi,
My configuration is SipPhone<-->*1<--->*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
there is no audio.
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
I also tested chan_ooh323, but to fail.
Can anyone recommand best
2009 May 06
0
problems in h323 channels
Hi, all!
when my h323 phone dial in Asterisk system, i can hear nothing. and
the following is the log slice i picked from /var/log/asterisk/full.
ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1,
pwlib_v1_11_0, openh323_v1_19_0_1.
Best
Regards!
81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection
81949 [May 6 10:07:51] VERBOSE[29627] logger.c:
2005 May 19
0
Re: Asterisk-Users Digest, Vol 10, Issue 154
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
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http://lists.digium.com/mailman/listinfo/asterisk-users
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asterisk-users-owner@lists.digium.com
2004 Oct 05
0
H.323: Inbound calls, incorrect remoteIpAddress
Hello,
I'm running Asterisk 1.0.1 (the same was with 0.9, 1.0). When it
receives inbound H.323 call it makes connection and uses local
127.0.0.1 address to send audio stream:
remoteIpAddress: 127.0.0.1
When making outbound calls from Asterisk it makes correct connection
to send audio stream. Is it a bug in h.323? Is there some more
settings to make in .conf files?
See detailed debug below:
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 Dec 22
1
Asterisk->AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
My configuration is:
[ISDNPRI] -- [CISCO AccessServer AS5350] --<H.323>-- [ASTERISK] --
[CISCO ip phone 12SP+/Skinny]
When call is initiated from IP phone -> Asterisk -> AS5350 -> ISDN
everything working ok (RTP is ok).
But, when call coming from ISDN -> AS5350 -> Asterisk -> IP phone
IP phone party can hear ISDN party, but ISDN (incoming) party canNOT
hear IP phone party
2004 Dec 22
0
Asterisk->AS5350 misplaced RTP to 127.0.0.1(AS5350 party don't hear)
Try sending 5350 config and oh323.conf, versions, etc...
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Goran Dj.
Sent: Wednesday, December 22, 2004 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk->AS5350 misplaced RTP to
127.0.0.1(AS5350 party
2003 Dec 16
1
asterisk and cisco call manager via h.323
Does asterisk work with CCM as gateway ?
When I trying call asterisk,I totally can't hear any sound.
When call ohphone - works good.
10.0.1.219 is CCM, 10.0.1.207 asterisk.
Trace messages here :
--------------------
== New H.323 Connection created.
-- Received SETUP message...
== Setting up Call
-- Calling party name: [5001,]
-- Calling party number: [5001]
-- Called party
2003 Sep 17
2
help jeremy
* compiled from cvs, i am trying call ip phones in callmanager 3.2
10.17.0.2 is my callmanager
i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2!
thereby causing no audio from * to ip phone. audio from ip phone to * is ok.
only callmanager calls fail. netmeeting works ok...
here is the debug, thanks for any info
~kelvin
H323 debug enabled
--
2005 Jul 03
0
H323 with GSM codec is not working
Hello,
I'm trying to use the GSM codec with Nufone H323 but it's not working.
Does somebody has some idea? Have I missed something?
Thanks!!
Celso Fassoni
Some additional info:
(I'm using CVS-HEAD - downloaded today)
monkey:~# cat /etc/asterisk/h323.conf
[general]
port = 1720
bindaddr = 192.168.0.100 ; this SHALL contain a single, valid
IP address for this machine
2003 Oct 13
1
chan_h323 - Segmentation fault (core dumped)
Hi all:
I've got some core dumps when I use chan_h323. I dial an extension using
h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes *
hangs, sometimes not. The client used for test es SjPhone
(http://www.sjlabs.com/).
This is the data for one core dump:
(gdb) bt
#0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790
#1 0x41f8879c in create_connection
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.
2005 Jan 28
0
Problems with H323/G729--No NATting and no Dynamic IP involved...
Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound at all nor the other side does (complete silence in both
sides). I thought this would just happen
2005 Jan 27
0
How can I check the selected codec for a call?
Hello... I'm having problems with H323/G729 setup. Below is the output
of h.323 debug when making a call. I use a SIP phone connected to an *
box in the same LAN. The * connects to a h323/g729 PSTN terminator
through internet. Calls rings and are answered in the other side, but I
get no sound at all nor the other side does (complete silence in both
sides). I thought this would just happen
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:
2007 Feb 14
0
Asterisk & CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323.
Cisco conf:
dial-peer voice 8 voip
destination-pattern 2...
session target ipv4:<asterisk ip>
codec g711alaw
no vad
h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;
disallow=all
allow=alaw
allow=ulaw
allow=gsm
context=from-internal
extension.conf
[from-internal]
exten =>