;
and right before
if (c) {
need to analyse result of wait:
if (sched) ast_sched_runq(sched);
if (ms==0) retry = AGI_NANDFS_RETRY;
that's the only changes reqired to get correct background work with FastAGI().
--
Regards,
Anton Fedorov
Call2ru service
E-Mail: datacomp...@call2ru.com
Hi!
I've came to a requirement to have over 256 /dev/zap/x
device files (512) to have 16E1's accessible (one of the
third party channel drivers - chan_ss7) addresses old-style
and needs this device files).
Any advice how to do this possible highly appreciated!
Thanks in advance!
Anton
and
zaptel.
Matthew Fredrickson
On Jan 29, 2007, at 4:43 AM, Anton wrote:
I've got a segfault of the 1.2.14 zaptel 1.2.12 libpri
1.2.3
Loaded symbols for /lib/tls/libnss_dns.so.2
#0 0xb761a11e in q931_hangup (pri=0x8231eb8,
c=0x83705e8, cause=16) at q931.c:2879
2879if (c
to chose a context is very tasty
thing, making dialplanning easier in many aspects.
Regards,
Anton.
On 29 December 2006 15:50, Johansson Olle E wrote:
29 dec 2006 kl. 07.17 skrev Anton:
Guys,
Have anyone thought about implementing a kind of SIP
extension, to call a context
the header on the inbound
invite.
It's all something you can solve within the dialplan,
without changing the
source.
/O
2 jan 2007 kl. 10.56 skrev Anton:
Olle,
Now with IAX we can call any context on the destination
Asterisk server, like
Dial(IAX2/destAsterisk/[EMAIL PROTECTED])
I
(for
instance 256's node) Cannot open 256 No such device or
address (6) error. If someone could please give a clue how
to make them accessible, it will be highly appreciated. Is
it a linux limitation or just a variable in Zaptel sources?
Please help.
Thanks in advance,
Anton
so far, they are excellent in other means,
and if you don't have long E1's...
On 22 December 2006 23:19, Anton wrote:
Yes, It is equipped with
2xSangoma 8xE1ExpressPCI cards+1xSangoma 2xE1 card
making in total 18xE1 links. Yes Eighteen E1's.
I've tested it in PRI mode Back-to-back (with ALL
of DEVFS?
2006/12/23, Matthew Fredrickson [EMAIL PROTECTED]:
On Dec 22, 2006, at 11:49 AM, Anton wrote:
Guys,
In a system with more than 8xE1 it's impossible to access
to /dev/zap/(channel no) channels higher than 255. ZAPTEL
Makefile generates only 255 instances and adjusting
Makefile
I'd like to add that an existing feature when you do
attended ZAP transfer, and than press flash the second
time - transfer becomes a conference. it's just nasty. Any
office PBX (Panas for example) do a conference while
pressing 3 (or conf on proprietary extension) - then
conference occurs.
Of Anton Sent: Thursday, November 23, 2006 10:25 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] IAX2 very CPU hungry
On 23 November 2006 01:13, Martin Vít wrote:
Anton wrote:
Hi!
Just changed the protocol when P4-3Ghz (1.2.12.1) box
have been dying receiving 60-IAX2
PC's talks to each other, so there must not be lots of
registrations. Lots of calls, yes, THey both have minimum
2xE1 PRI with traffic they exchange via IAX or SIP
On 25 November 2006 09:16, Steve Kann wrote:
Instead of speculating (registrations, some error in the
call setup, etc), why
I have to disagree with you. Seems you have never woked much
with CISCO or similar equipment CLI. Otherwise you would
not argue. Don't reinvent the wheel. There is ony one con-
according to the Luigi - it's not easy to implement it now
in the current code.
On 19 November 2006 13:53, Tzafrir
would be just great to have it.
On 18 November 2006 04:20, Bob Atkins wrote:
Please let me 2nd, 3rd and 4th this request and add an
emphatic /please!/ Such a change would result in a
/_much_/ more efficienct CLI that functions in much the
same way that Cisco's IOS CLI works and that many of
Hello,
I'm starting getting regillary a segfault on the 1.2.12.1 PC
Should I fill the bug report with the following data, or that already have been
resolved in 1.2.13?
Load is quite high. 8xE1 - PRI + IAX
---
Core was generated by `/usr/sbin/asterisk -vvvg -c'.
Program terminated with signal
P4D,3.0G,1024M
max load I've seen according TOP - 56% of asterisk process
On 18 November 2006 21:10, you wrote:
8 E1s?
What is machine spec?
Isamar
On Sat, 18 Nov 2006, Anton wrote:
Hello,
I'm starting getting regillary a segfault on the
1.2.12.1 PC
Should I fill the bug report
. The length is plausible for a Meta frame carrying
a single (G.711) packet.
Details in the bugtracker.
On 2 November 2006 14:10, Tim Panton wrote:
On 2 Nov 2006, at 04:31, Anton wrote:
Again, the OLD issue - after a while - IAX becomes
1way-or-no-audio operation. Any suggestion or anyone
But where is the mentioned code? Could you please post a
link, where to get it from?
On 5 September 2006 07:47, Kannaiyan Natesan wrote:
Hey,
Is this code released by Digium?
Looks like directly from digium. Is it GPL with License
and Royalty?
Unlimited channels and no restriction
It's not binded with Asterisk, but I'm behind a satellite
connection, and regullar faxes works - but with CISCO
equipment...
On 28 July 2006 20:23, John Lange wrote:
On Fri, 2006-07-28 at 09:08 -0400, Andrew Kohlsmith wrote:
Faxes are more of a problem because their lower-end
connection
.
when happens, asterisk does not shutdown itself on stop
now - just silently continues operation - ihave to kill -9
it.
Does anyone have similar behaviour? Any suggestion what
types of logs I have to look/swith on to report in more
detail?
Regards,
Anton
appreciate any help!
Regards,
Anton
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
Thanks for info guys! Somehow I've missed that posibility...
Was completely sure that there is no any extra options
after sip debug.
On 22 May 2006 10:48, Dinesh Nair wrote:
On 05/22/06 13:10 Anton said the following:
sip.conf debuglevel=x that only SELECTED phone would be
debugged
What does it mean?
If it's like denying overlapping calls from the same
originator to the same destinations at the same time - I
would be very interested to know how to do that also.
On 12 May 2006 17:47, Nicolas LEGROS wrote:
Hi !!
I'd like to configure my extensions.conf file in
order
It need for linux distributions, for updating third part
modules independent with asterisk.
There is a number of modules in fact, which are exist in the
binary form, and quite complicated to get them rebuilt or
updated to a new version, so if there is a way to keep
binary compatibility it
On 1 May 2006 17:52, Kevin P. Fleming wrote:
I'm not aware of any other open source projects that
attempt to preserve binary compatibility across major
release versions. Certainly it is not the norm, and is an
undue burden on the developers of the open source project
to have to maintain
I also thought the same until found that if I press buttons
on my Treo650 too fast, DTMF tones did not get recognized
by my Cisco IVR. I think there are different behaviours
with different phones.
On 24 April 2006 04:52, Steve Underwood wrote:
Vahan Yerkanian wrote:
Steve Underwood wrote:
Dan,
Is you version of ooh323 relatively stable?
May I have a code please?
Regards,
Anton.
On 21 April 2006 00:01, Dan Austin wrote:
Jeremy wrote:
Nicolas LEGROS wrote:
I'm currently working on the channel H.323 (from
Asterisk v.1.2.7.1).
Development really should be done from
Dan,
Could you please point me where is the SVN of chan_ooh323,
since I see on the SF i see only the ooh323c library, and
not channel (am I missing something?)
Thanks in advance!
Anton.
On 21 April 2006 03:23, Dan Austin wrote:
Is you version of ooh323 relatively stable?
May I have a code
Hi Dov,
What that corrections you posted does exactly?
Sincerely,
Anton.
On 17 April 2006 18:46, Dov Bigio wrote:
Another change that was made on my Asterisk 1.2.6 and
helped a lot...
-
---
+++ res_features.c
Production asterisk is dying once in 24hours. Any clue anyone?
#0 alawtolin_framein (pvt=0x825e120, f=0xb7a2c804) at codec_alaw.c:173
173 tmp-outbuf[tmp-tail + x] = AST_ALAW(b[x]);
(gdb) backtrace
#0 alawtolin_framein (pvt=0x825e120, f=0xb7a2c804) at codec_alaw.c:173
#1 0x08069e74
The following backtrace I've got. Is it known problem?
* Load is tens of thousands calls per 24 hours.
#0 alawtoulaw_framein (pvt=0x8205648, f=0x8169554) at codec_a_mu.c:128
128 tmp-outbuf[tmp-tail + x] = a2mu[b[x]];
(gdb) backtrace
#0 alawtoulaw_framein (pvt=0x8205648,
Guys,
there is an implementation of the application independent
jitterbuffer, with abstraction etc, can anyone say could it
be integrated with asterisk RTP channels?
http://www.speakup.nl/opensource/jitterbuffer/
___
--Bandwidth and Colocation
On 20 March 2006 06:56, Paul Cadach wrote:
Asterisk ALREADY HAVE 3 (!!!) different channel drivers
for H.323. Two of them like MVTS based on OpenH323...
WBR,
Paul.
All segfaults for me in some conditions. I need a stable
implementation. Though there was a guy in IRC telling that
he's
Charles,
Don't you think that is a question to asterisk-users list?
On 20 March 2006 09:00, Charles Huang wrote:
Hi,
I am currently integrating our company's Toshiba PBX with
the Asterisk version 1.2.1.
I bought Quad T1 card, and making the port 1 to connect
to PSTN PRI (use pri_cpe in
Matt Riddell wrote:
Anton Tinchev wrote:
Will be there new card?
I'm asking it, 'couse i'm going to buy 3-4 cards?
Or i should wait for the new one?
I'd buy now. There has so far been no information on the new cards from
Digium (I.E. no confirmation that the card will exist and consequently
(2000-5000 users for 2 years) shortly after New Year.
I evaluating using of * for this deployment. Unfortunately, our team didn't include
expirience C programmers, so i think that we will need some external support in this
field.
Thanks.
Anton Tinchev
Unix Solutions ltd
Sofia, Bulgaria.
P.S
35 matches
Mail list logo
