alTime:
Delete SQL: DELETE FROM channels WHERE uniqueid = '1205172794.6453'
[Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime:
Deleted 1 rows on table: channels
[Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166'
removed from call limit 8
[Mar 10 11:1
On 3/9/08, Godwin Stewart Horwich IT Services <[EMAIL PROTECTED]> wrote:
> On Sun, 9 Mar 2008 16:22:35 +0200, "Atis Lezdins" <[EMAIL PROTECTED]> wrote:
>
> > I think that giving 's' argument should silence all prompts including
> > auth-thankyou.
thankyou.gsm.
>
I think that giving 's' argument should silence all prompts including
auth-thankyou. You should report a bug on http://bugs.digium.com ,
fixing this should be trivial.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: ati
track of connections - connect and disconnect in dialplan.
Or there's odbc module that creates permanent connection, and allows
you to declare SQL functions with replacable variables, but personally
i don't like having additional layer. For this you can search
mailinglist, it&#
ool if you want to drop .call files
load => pbx_spool.so
; Functions
load => func_strings.so
load => func_rand.so
load => func_logic.so
load => func_cut.so
load => func_callerid.so
; Resources
; if you don't have agi, you may remove th
here are probably not that difficult to script within
> Asterisk itself, or with a simple wrapper.
>
> Test of audio quality is something I'm not really sure how to do.
Run tests, and ChanSpy() them? See at which point decrease of quality
becomes hearable.
Regards,
Atis
--
Ati
very good resource for that -
http://lartc.org/
Also i found this yesterday, could be good for start. It doesn't
assume fixed bandwidth, but just gives priority to VoIP packets.
http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk
Regards,
Atis
>
> > IAX is mor
Well, PHP is language in which i'm coding most for last 5 years, so
when i needed something fast, i took it. And maybe some day it will
have web interface.
Regards,
Atis
On 2/19/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
> Just out of curiosity, why PHP?
>
> Atis Lezdins w
m agent call every 10-20
seconds, and random customer call every 20-30 seconds), all in PHP
with some functions and objects to make interaction easy.
Regards,
Atis
>
> Atis Lezdins wrote:
> > On 2/18/08, Khaled Chehab <[EMAIL PROTECTED]> wrote:
> >>
> >>
>
iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's
designed for testing queue-agents scenarios but i'm sure you can
adapt.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
n by
# svn switch http://svn.digium.com/svn/asterisk/tags/1.4.19
and re- make, make install
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--
PROTECTED]
> http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
>
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>http:
hat shows you those states
by groups, etc..
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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with Yahoo! Search.
> http://tools.search.yahoo.com/newsearch/category.php?category=shopping
>
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> To UNSUBSCRIBE or update options
(even if it returns ok
exit stats that means shutdown), and then stop asterisk by "stop when
convenient")
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
,100,my_field)=foo);
Also, seems that func_realtime will soon support SQL INSERT's and DELETE's :)
Regards,
Atis
>
> Atis Lezdins wrote:
> | On 2/13/08, Rob Hillis <[EMAIL PROTECTED]> wrote:
> |> -BEGIN PGP SIGNED MESSAGE-
> |> Hash: SHA1
> |&
On 2/13/08, Rob Hillis <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Atis Lezdins wrote:
> | By RealTimeUpdate do you mean func_realtime? It shouldn't care, as
> | cache is not implemented in realtime level, but higher (chan_sip).
p.c - that
way you will gain some extra performance, as there's no manager/cli
overhead.
However if you're uncomfortable with C, the app_system shouldn't cause
any troubles :)
Regards,
Atis
>
>
>
> Atis Lezdins wrote:
> On 2/12/08, Rob Hillis <[EMAIL PROTECT
e any suggestions?
You could write dialplan application to do the same in chan_sip. Code
should be very simple, just the processing of one argument and reusing
existing functions. If you'll argument good enough why you need it, i
think it could be included in asterisk.
Regards,
Atis
-
es, and you can
write few of them in extensions.conf - it's a small trouble when
compared to gain from separation of different functionality.
Regards,
Atis
>
> I'll try to remeber it for next time :)
>
> Cheers,
>
> Yves.
>
> On Fri, 2008-02-08 at 14:36 +0200, A
r data that actually changes.
Regards,
Atis
>
> * Using numbers instead of 's' => already tried, no changes
>
> * Renaming contexts without underscores => tried it right now, no
> changes
>
> Thanks for all those ideas.
>
> Yves.
>
> On Thu, 2008-02
gt; Regards
>
Sure:
context dial-out {
_X. => {
if ("${GROUP_COUNT(gw2)}">"${GROUP_COUNT(gw1)}") {
Set(OUTBOUND_GROUP=gw1)
Dial(SIP/[EMAIL PROTECTED])
} else {
Set(OUTBOUND_GROUP=gw2)
Dial(SIP/[EMAIL PROTECTED])
}
}
}
Regards,
Atis
--
ble. Try a:
> >
> > SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0'
> > AND priority='1'
> >
> > If that fails, you have your answer.
> >
What version? You could try replacing pipes with commas. Do you have
tion. Add ResetCDR(w) directly before dial.
2) First CDR for whole call (including Dial to operator and anything
before) and second for just Dial to operator - use ForkCDR() before
Dial.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ L
BOSE" level and verbosity
set to 3. You can just uncomment "full" line in logger.conf and issue
"core set verbose 3" in CLI.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7
en I use this dialplan and during the intro.3gp I press the #-key the
> call will be ended.
> But I got three different CDR's.
>
> Does anybody know how I can use one CDR instead of 3 different CDR's
>
> Kind Regards,
>
>
> Arjan Kroon
> Mobillion BV
>
> -
ncomment the "full" line in logger.conf) that
says anything about posting CDR and previous few commands would be
useful.
Regards,
Atis
>
>
>
> Kind Regards,
>
>
>
>
>
>
>
>
> ___
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;
If ("${DIALSTATUS}"="ANSWERED") {
break;
}
Wait(1); // wait so that this loop doesn't eat much CPU.
Set(remaining_time=${EPOCH}-started+120);
}
if ("${DIALSTATUS}"!="ANSWERED") {
Busy();
}
Regards,
Atis
--
Atis Lez
or two underscores. For
example - if you want to have one common unique identifier for call -
do this at beginning of each channel:
if ("${call_id}"="") {
Set(__call_id=${UNIQUEID});
}
Then you can use this to store something shared in asterisk internal DB()
Regards,
Atis
--
7;t.
Retrieve the variable from inside AGI.
http://www.voip-info.org/wiki/view/get+variable
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--
agents channel.
I think it's correct - if you set userfield - it will be written - no
matter if call has been answered or not.
If you would do ResetCDR before entering queue, you would have
disposition=ANSWERED for queue calls that got answered by agents. Also
you would have billsec= agent t
> others having
> > the same problem it will fade away.
> > http://bugs.digium.com/view.php?id=11849
> >
> > From my point of view the design solution to this problem would be
> > as simple
> > as changing the CDR generation from one CDR per bridge to generatin
-
>
>
> Thanks a lot,
> Thomas
>
> --
> Thomas Kenner
>
> _______
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>
> asterisk-users ma
, whereas, I need to split the call
> after it is established based on a DTMF stimulus. Are there any other
> ways of accomplishing this goal?
>
> Any thoughts, ideas?
>
> Thank you,
You should take a look at this:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
See the appli
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>
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROT
f 3GHz and 8Gb RAM (2 of which are used by MySQL indexes).
Asterisk is running on same machine.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
__
rds,
Atis
>
> Thanks,
>
> Jaap
>
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--
Atis Lezdins
On 1/17/08, Atis Lezdins <[EMAIL PROTECTED]> wrote:
> On 1/17/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> > On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote:
> > > Hi,
> > >
> > > I'm wondering why zttest shows
> > > B
kported patch for 1.4
Regards,
Atis
>
> thanks
>
> --
> Cyril SCETBON
>
>
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On 1/17/08, Mark Michelson <[EMAIL PROTECTED]> wrote:
> Atis Lezdins wrote:
> > Hi,
> >
> > I'm wondering - why SIP device state doesn't get updated to anything
> > else, except Not In Use.
> >
> > For queue call (with Local channel) i get:
em successfully working. Sometimes they crash and
needs reboot, but generally they are ok.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
xx.xxx.xxx.xxx
regexten:
cancallforward: yes
setvar:
Any help would be appreciated.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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>
";
#include "extensions_ivr.ael";
#include "extensions_globals.ael";
However i'm using
#aelparse -d -n -w -q extensions.ael
to dump ael into .conf format
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
On 1/17/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Thu, Jan 17, 2008 at 03:09:59PM +0200, Atis Lezdins wrote:
> > Hi,
> >
> > I'm wondering why zttest shows
> > Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469
> >
> &
46272 0
wcfxo 21536 0
wctdm24xxp121024 0
wcte11xp 34848 0
wct1xxp23456 0
wct4xxp 323904 96
zaptel202984 215
zttranscode,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2
crc_ccitt 10817 1 zapte
s not
Originate'd from manager.. Btw, why not using app_meetme, bundled with
Asterisk? Some time ago i was working on similar solution - sending
existing call to meetme and adding another Playback() to both calls,
however i currently don't have working version of this. You can try my
sample sce
te something like this (in
realtime of course)
[from-queue]
exten=>200,hint,SIP/300
exten=>200,1,Dial(SIP/300)
Would this send update to queue of SIP/300 state? All the RINGING/INUSE?
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
s will be available from Realtime.. Managing more
than 50 users makes static config a nightmare, and AFAIK there is no
ways how to create hints with variables/extension masks. So, it is
logical to ask for hint support in Realtime.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
s-1.4/ ),
however it's just being tested for 1.4, and probably will be available
only in 1.6.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
{tv_sec = 1101244224, tv_usec = 5241072}, ru_maxrss =
469252700480, ru_ixrss
status = 0^
__PRETTY_FUNCTION__ = "ast_safe_system"^
...
Seems that asterisk got locked on fork() and that's quite strange.
Other threads shouldn't have any effect of it. Any ideas what might
have happene
;2
- to accept only two simultanous calls per agent.
And finally for outgoing calls - you just don't check any GROUP_COUNT,
just set the group - that will allow any number of outgoing calls.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs
Vinicius Fontes wrote:
> Hey guys, I don't know if this is the right place to ask this. I was
> thinking about reporting a bug, but maybe it's better to sort out if
> this is really a bug or just me being lame.
>
> I want to record *every* call in my Asterisk box, so I use the
> MixMonitor()
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--
Atis Lezdins
VoIP Developer,
IQ Labs
able. This would require use of Agent or Local
channel, so you can do custom Dial for queue member. Works for me.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
Remco Barendse wrote:
> I have an analog GSM Gateway that is connected to a normal SIP ATA device.
>
> Basically what it does is this : when you call the extension nr. of the
> SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
> dial tone, and then dials whichever DTMF to
rns nothing even if I try to do a query like
> "SELECT 1;"
>
> Is anyone in the same troubles ? Do you advice me another solution to
> connect to my database ?
app_addon_sql_mysql from asterisk-addons - it works fine for me.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Lab
ht be not so
intuitive to configure, but it really helped us to find memory
corruption problems under high load.
Please see:
http://lists.digium.com/pipermail/asterisk-users/2007-November/200429.html
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
ease have had some serious changes that blocked our
planned upgrades - for example some memory corruption that raised
between 1.4.10 and 1.4.12 that was very hard to track down. This shows
that having 1.4 in bugfix-only state is not actually working that good -
we have to test each new release
Tony Plack wrote:
>>> All I can say is with 1.6, if a change is made that causes
>>> something that worked in 1.4 not to work in 1.6, please think
>>> twice, three times or four times before making the change, or
>>> making the change in such a way that it won't break dialplan
>>> stuff from 1.4.
>
d I assume that you have the
> operator=yes in voicemail.conf. I would look to these sections to determine
> what is occurring.
>
> Somewhere in your config you have the system trying to call xxx:[EMAIL
> PROTECTED] for the operator.
Btw, the warnings are for IAX channel. D
> -- Playing 'vm-intro' (language 'en')
>== Spawn extension (default, 10100, 4) exited non-zero on
> 'SIP/10100-082244c0'
> --
>
> Can anyone help me with this? I want immediately voicemail answer... maybe
> these error is the cause... I saw that in this pause t
Benjamin Jacob wrote:
> Hello ppl,
> Am totaly new to this zap thingy.. zapped I would say I am! (couldn't
> resist that cliche...).
>
> Just like sipp for testing SIP channels, do we have any such tools to
> test zap channels?
>
You can try PBX Testing Framework. It's using SIP by default, bu
graceful shutdown; perfect
> for routine system maintenance tasks on production
> servers handling continuous traffic.
stop gracefully
"help stop" shows other similar options
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone:
Benny Amorsen wrote:
> The multiple registrations issue is currently our largest problem with
> Asterisk. The workarounds are horrible and complicated. Say we want 3
> phones ringing when a new call comes in, but if just one of them is
> busy outbound or inbound, they should all return busy for inc
Anciso, Roy wrote:
> Is there a way to tell asterisk to beep every few seconds rather than
> play MOH.
I suppose you mean the beeps you hear when dialing any phone number -
the ringing indication? I'm not sure that this would work with queues,
however it works great with Dial(). I'm also not su
Peter Pauly wrote:
> I currently have the following setup:
>
> exten => 2000,1,Playback(/var/lib/asterisk/sounds/Greeting)
> exten => 2000,2,Queue(Qabcdef|t)
> exten => 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy)
> exten => 2000,4,Hangup
> exten => 2000,103,Hangup
>
> What happens is,
Mike wrote:
> Hi,
>
> The Asterisk Wiki (page:
> http://www.voip-info.org/wiki/view/Asterisk+func+cdr) mentions I can set
> any custom CDR field I want. Here is the example it gives:
>
> ; Update our accountcode field and then save some random music facts too
> exten => s,1,Set(CDR(accountco
1:49:44","14","5","ANSWERED","DOCUMENTATION","","1196790570.4259",""
>
> If I don't answer, I still get an 's' CDR with No Answer. Any ideas how
> to stop that? Thanks.
>
> Peder
>
>
>
Pepo wrote:
> Hi friends.
>
> I have problems with the voicemail system, when some user "forward" the
> message to other box all the Asterisk falls down and restart.
>
> How do I disable the option to forward messages in voicemail (option 8 in the
> menu)? and Which can be the cause for the pro
Steve Underwood wrote:
> Darryl Dunkin wrote:
>> Does anyone have any opinions on the music on hold quality over G729?
>> The stock files seem to sound terrible over it, this is enhanced further
>> by calls coming from the PSTN via a Zaptel gateway. I am only using the
>> stock wav files and have n
> # Jesse Molina
> # Mail = [EMAIL PROTECTED]
> # Page = [EMAIL PROTECTED]
> # Cell = 1.602.323.7608
> # Web = http://www.opendreams.net/jesse/
>
>
>
> _______
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Yehavi Bourvine +972-8-9489444 wrote:
> Hello,
>
> I would like to check whether an extension is busy or not before calling the
> Dial() application to it (for example - to play a Busy if it is on
> conversation).
>
> How do I check it? In the trunk version there was a function
> DEVSTATE(SIP
busy
> >> exten => s,n(all-busy),Hangup
> >
> > Regards,
> > Atis
> >
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> >
> > asterisk-users mailing list
> > To UNS
Jakub Syrek wrote:
> I thing there was an error in last version of my macro, correct one (i
> hope):
Just test it :)
>
> [macro-call]
> ;sip1 - firs channel from sip outgoing cals operator
> ;sip2 - second channel from sip outgoing cals operator
> ;sipn - N channel from sip outgoing cals operat
Jakub Syrek wrote:
> Hello.
> Is it possible to check if SIP chanell is busy in asterisk?
> I have N accounts from my provider and i can dial only one call per account.
> I wanto my asterisk to check if first acount is busy, if yes try second and
> so on..
> I was wondering if ChanIsAvail will sui
rs
> >
>
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nik600 wrote:
> Hi
>
> i've read this post
> http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html
>
> I just want to know if there are some upgrades... on 1.4 or 1.2.
>
> I'd like to store two records in the CDR instead of one, when a call
> is transferd.
>
> Is it possibile now?
Dan Casey wrote:
> Sorry for my very delayed response. To answer a few questions:
> 1. Right, the *ANI*DNIS* is not working correctly. When the telco sends
> it, we are always missing the beginning of it. I almost always get a 7
> digit ani, but sometimes it is 8 or 6.
I won't be able to help w
On 11/19/07, Brian J. Murrell <[EMAIL PROTECTED]> wrote:
> On Mon, 2007-11-19 at 16:26 +0200, Atis Lezdins wrote:
> > On 11/19/07, Brian J. Murrell <[EMAIL PROTECTED]> wrote:
> > > I am using asterisk 1.4.10 and seem to be having a problem with blind
> > &g
Baji Panchumarti wrote:
> On Nov 19, 2007 10:13 AM, Atis Lezdins wrote:
>
>> Tilghman Lesher wrote:
>>> On Monday 19 November 2007 02:45:17 Abdul wrote:
>>>> Hi all,
>>>>
>>>> I was playing with asterisk .gsm sound file to work for call
Tilghman Lesher wrote:
> On Monday 19 November 2007 02:45:17 Abdul wrote:
>> Hi all,
>>
>> I was playing with asterisk .gsm sound file to work for callback.
>>
>> But the quality is very poor and sound is very low so we cannot clearly
>> hear what is sound played.
>>
>> Is there any option in aster
nnel 'Zap/1-1' jumping out of macro 'dialhouse'
> Calling party hears click:
> -- Hungup 'Zap/1-1'
>
> And the call is dumped by that point.
>
> As a side note, the timeout between the '#' and the extension is quite
> short. 3secs or so it seems. Any way to increase that?
>
> Cheers,
> b.
&
fine. You should provide error messages you got.
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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. So -
take a look at sipp, etc, but afaik - there is no match to real-life
call load simulation.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
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-
Stefan Guenther wrote:
> Hello,
>
> I have a strange situation:
>
> I can talk to other SIP phones and via ISDN to the outside, but I don't hear
> playbacks or the voicemail messages.
> Asterisk show in the cli, that the corresponding files are played, but I hear
> nothing at all.
>
> Here is as
e sending call to queue. It would reset
call's answer status in CDR, so call would have queue's answer status.
Also you might be interested in reverting this patch:
http://lists.digium.com/pipermail/asterisk-users/2007-October/199532.html
Regards,
Atis
--
Atis Lezdins
Thomas Stein wrote:
> Hi.
>
> Asterisk 1.4.12.1.
>
> I get a lot of message like this. Someone knows what this message mean? Do i
> have to worry about it?
>
> [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on
> channel 'Local/[EMAIL PROTECTED],1' not posted
> [Nov 7 15:24:25] N
> I am just wondering - it there any way how to protect a conference from
> being abused by someone?
> I know I can request pin, but that pin is then hardcoded in meetme.conf
> and normal user can not change it.
>
> I would like to establish an admin user who could set a pin for the
> conference t
or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
I'm just annoyingly copying URL and deleting first part of it - it's
simpler that to quote whole URL.. And without quoting - bash can't
correctly interpret the ? stuff..
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAI
Rilawich Ango wrote:
> Hi all,
>
> Recently, I have upgraded the asterisk as following.
> asterisk-1.4.13
> asterisk-addon-1.4.4
> libpri-1.4.1
> zaptel-1.4.5.1
> Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh
> After upgrade, the server get segfault randomly an
You have to revert patch for issue http://bugs.digium.com/view.php?id=10659
More specific - that is - remove those two lines from main/cdr.c
if (cdr->disposition < AST_CDR_ANSWERED &&
(ast_strlen_zero(cdr->channel) || ast_strlen_zero(cdr->dstchannel)))
continue; /* people don't want to se
Douglas Garstang wrote:
> I'm confused about something.
> It's the way Asterisk handles the A leg (ie the first party dialed) on
> an originate command via the Manager Interface.
>
> Lets say our originate commands looks like this:
>
> ACTION: Originate
> Async: yes
> Timeout: 6
> Exten: call
file is too big and asterisk takes too much time
> reloading the entire file.
>
> Is there a way to avoid this probleme or another way to add/remove sip
> phones dynamically ?
Realtime?
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
Regards,
Atis
--
Atis Lezdins
VoIP D
mmands?
Yup, it's also a problem for me, but it haven't ever crashed server. It just
makes specific remote process unresponsive. There's a patch for 1.4, but i
guess it wouldn't be hard to backport it for 1.2
http://bugs.digium.com/view.php?id=10847
you might also want the o
)
> but am still getting the same error. The first step in that
> email-hungup macro is being executed and nothing else.
>
> Also, this:
>
> Oct 17 22:14:54 -- Executing [EMAIL PROTECTED]:2] GosubIf("Zap/3-1",
> "1?notify|1") in new stack
This means, the variable
tion
> is throwing a parse error, I'd greatly appreciate your insight. This
> was really frustrating and is probably a stupid mistake.
Try removing spaces around =
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +37
entical). extensions.conf is generally the same format as other
config files, however you should check syntax of applications, and priority
ordering (if you are using it at all).
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: at
forever.
I recently had the same problems. Just that wasnt't related to installing any
new hardware. You can check out the issue
http://bugs.digium.com/view.php?id=10775
Could you provide your OS and glibc version? Also - can you try to disable
IAX?
Regards,
Atis
--
Atis Lezdins
VoIP Deve
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