Displaying 20 results from an estimated 8000 matches similar to: "channel.c: Avoided initial deadlock"
2007 Sep 20
10
IAX Java Softphone?
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded in some other app/let. Preferably GPL. Thanks.
--
(C) Matthew Rubenstein
2006 Dec 12
3
outgoing call on ISDN PRI
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered on
the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and how
to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
Thanks
2007 Apr 05
5
Open Source VoIP client (on a webpage)
I need to decide on the best way to add a voip SIP or IAX client to a
website. I'm thinking that I'd like it to be inline, like an aplet, on
the page. I've got some asterisk servers running to connect up to, so
the real challenge is finding an easily integrated open source client.
Any suggestions from those who know?
Jason
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2006 Jun 14
3
WRTG54GS Capacity
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and the
WRTG54GS connects to a remote Asterisk server using IAX2 trunking
using GSM codec.
Thanks,
Daniel
2007 Jul 03
4
Google acquires Grand Central
Ooops did Google just become a carrier :)
http://googleblog.blogspot.com/2007/07/all-aboard.html
I hear stocks crumbling worldwide as I type.
Cheers,
Dean
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2007 Jul 23
2
IAX Encryption
I am playing around with IAX encryption and have had good success.
I read somewhere, that trunked packets are not encrypted. Does
anybody know if this means the trunk packets themselves are not
encrypted but the voice frames in them are encrypted or does this
mean that if you are using trunking then encryption of the voice
frames will not occur. I have used Wireshark to sniff the packets
and it
2007 Aug 30
0
WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries!
Hello!
I clear remarks in Makefile:
DEBUG_THREADS = -DDEBUG_THREADS -DDETECT_DEADLOCKS
But same things in CLI:
Aug 30 18:16:31 WARNING[22292]: channel.c:780 channel_find_locked:
Avoided initial deadlock for '0x82f2fe0', 9 retries!
-- Zap/32-1 is proceeding passing it to Zap/31-1
-- Zap/32-1 is ringing
-- Accepting call from '2177' to '7141278' on channel
2006 Apr 01
1
channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!
I never so this error.
I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
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2007 May 04
1
ASA-2007-013: IAX2 users can cause unauthorized data disclosure
> Asterisk Project Security Advisory - ASA-2007-013
>
> +----------------------------------------------------------------------------------+
> | Product | Asterisk |
> |----------------------+-----------------------------------------------------------|
> | Summary | IAX2
2007 May 04
1
ASA-2007-013: IAX2 users can cause unauthorized data disclosure
> Asterisk Project Security Advisory - ASA-2007-013
>
> +----------------------------------------------------------------------------------+
> | Product | Asterisk |
> |----------------------+-----------------------------------------------------------|
> | Summary | IAX2
2006 Oct 11
1
1.4 beta2 on intel mac
Has anyone built and run asterisk 1.4 beta2 on an intel mac?
Did it work?
I've got it building ok (once I installed Xcode, wget and bison)
However Asterisk hangs on startup (halfway through loading the modules).
I have not (yet) had time to debug it, but I wondered if anyone else had
done this before me ?
Tim Panton
www.mexuar.com
2006 Nov 13
3
FW: Desktop integration
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000066">
Hi Dean,<br>
<br>
I will check that site - thanks for the hint.<br>
The biggest problem I see with
2006 Mar 19
3
Annoying Asterisk Realtime Limitation
Well, this is a major pain in the ass.
I got realtime static working for sip.conf. 'Great!' I thought. That was until I realised I couldn't use it.
Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk doesn't like to use an interface name for it's bindaddr setting, so you have to put the IP address of lo:1 in there. If you put in 0.0.0.0, it seems to
2006 Apr 20
2
Cubix Softphone + Asterisk 1.2.6
I've tried Idefisk and Cubix Softphones, and they both work fine, except
for two issues:
1. Idefisk seems to have a longer delay between the time I can hit
tones, and
2. Cubix, while can send DTMF faster, never actually connects to an
Asterisk-dialed call -- I can't hear the party who answers.
#2 has been asked but unanswered here:
2007 Jan 27
1
Via EPIA channel_find_locked: Avoided initial deadlock
In asterisk 1.2 branch SVN 51363
zaptel svn 1980
libpri svn 393
addons svn 332
My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a
tdm400p (4fxo).
A call comes from zap, a SIP ulaw receives the call, talks for a while
and when SIP users tries to park the call, then dozens of...
WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial
deadlock for '0x91bb840', 10
2006 Dec 06
0
Avoided initial deadlock asterisk v 1.2.12.1 SIP clients IAX2 termination.
Periodically (as in sometimes several times a day and sometimes never)
I get A channel.c: voided initial deadlock for '0x82*****', 10 retries!
The ***** figure is different each time.
When this happens an active call (in or out) is dropped.
The setup is as follows:
handset --SIP--> Asterisk 1.2.12.1 --IAX2--> Terminating supplier
(don't know which software they are running).
2008 Jan 09
1
Help! channel_find_deadlocked: Avoided initial deadlock for ...
Hope someone can help.
I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.
Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial
2007 Jun 20
1
Res: Record CDR in a Oracle database
Hi All,
Thank's for your hint Tim Panton
I could connect my asterisk machine to my oracle machine.
I used unixODBC-2.2.11.tar.gz,
oracle-instantclient-basic-10.2.0.3-1.i386.rpm,
oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from
www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my
asterisk machine.
I can connect to my oracle machine with isql and in
2007 Jan 16
1
Ring tone too loud on IAX channel
Hi,
We are using MozIAX as a softphone with a USB headset and are making
outbound calls using IAX with ulaw encoding to our voip provider.
We're running asterisk 1.4
Users are complaining that the ring tone generated by asterisk is much
louder than the voice call once connected. They are having to turn the
volume down to avoid being deafened by the ring tone, but then have an
unacceptably