a lot, Dmitry.
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will be in the future
3) To use odbc to access mysql? but I could not find a procedure for it. And I
doubt it is possible.
BR,
Dmitry Pavlenko
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New to Asterisk
for queue_log
So I chose 1).
From: Lenz Emilitri lenz.lo...@gmail.com
To: Dmitry mbike200...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Thursday, November 22, 2012 3:36 PM
Subject: Re: [asterisk
if I need to authorise my outgoing calls
(probably sip.conf will be more logical in the future 12th version).
Hope this helps.
Dmitry Pavlenko
From: Ken D'Ambrosio k...@jots.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi, List
My SIP provider requires no authorization in incoming calls to my asterisk
11.1.0 box.
I was sure previously that insecure=invite,port disabled authorization
request during incoming calls to asterisk.
But today I tried to connect to a provider (which has MERA MVTS) but could not
Use tcpdump utility, capture all udp packets for the faulty t38 fax call.
T38 works reliably since asterisk 1.6.
Wireshark utility shows t38 signals so it is possible to debug it.
From: Eric Wieling ewiel...@nyigc.com
To: brya...@zktech.com
to a server with
Aqua software which compared this file to its original. then the quality
(measured in percents) were sent to Zabbix monitoring. actually this data was
used for analisys and it compares two files (not realtime).
BR,
Dmitry Pavlenko
From: Lenz
this helps
Dmitry Pavlenko
From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 23, 2013 11:25 AM
Subject: Re: [asterisk-users] Asterisk Log rotate not working
About 4 years ago I made it (outbound faxing):
I wrote a Perl script which fetched the message with PDF file from an external
POP3 or IMAP box. (I did not succeed to use a Fetchmail utility on Centos so
had to write my own one)
Then I wrote a BASH script which parsed the mail message, it took
Hi,
You also can see example script for create cluster.
https://github.com/netaskd/AFDINbeat
---
Dmitry Burilov
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent
algorithm to discover
any fraud attempts (I want to analize the security log as well).
BR,
Dmitry Pavlenko
On Fri, 10/18/13, binary dreamer binary.vor...@gmail.com wrote:
Subject: [asterisk-users] fraud detection
To: Asterisk Users Mailing List - Non
Hello!
Looks like h323 compiling is FAQ, but I didn't found an answer...
The same problem with 0.6.5 and 0.7.1:
gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE
-I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o
Hello!
Do somebody knows how to compile meetme2 with 1.0.6.
I readed wiki, applied patches, but no luck ;-(
Me be someone can give me working meetme2.c ?
:-)
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: (Each undeclared identifier is
reported only once
/usr/include/asterisk/lock.h:302: error: for each function it appears in.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Melekhov
Sent: Wednesday, March 16, 2005 4:36 AM
To: asterisk-users
Looks like there is no anoynomous access for this dir
:-(
Could you mail me this file?
Squid sent the following FTP command:
PASS yourpasswordand then received this
reply
Login incorrect.
- Original Message -
From:
Asterisk
To: Anil Kumar K ; Giovanni Powell
Cc:
we need for making it work?
Dmitry
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, and engineer around those numbers. Asterisk is no
different.
Thanks for advise. Unfortunately we don't have this kind of data for this
project. All we have is estimations.
Dmitry
Rich
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of them up (one at a time), and I'm still getting it.
What is this talking about?
probably you have incorrect permissions for /var/lib/asterisk/astdb.
It may be if you are running asterisk not under root.
I had the same problem. Changing permissions to rw solve it.
Dmitry
Tim
of using g729a codec from Digium.
I'm interested in resource requirements when codec translations performed.
Also its interesting to know how many simultaneous channels translated with
this codec and what kind of server hardware is used for this case.
Thanks,
Dmitry
to some number. When the call is finished he should be
able to press * and have possibility to enter another number he wish to
call.
- CDR should be written after call to each number.
Thanks,
Dmitry
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working properly.
Dmitry.
Any informations regarding the timeframe of appearance would be
appreciated...
Greetings
Bjoern
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to be transfered being disconnected.
Any ideas whats wrong?
Thanks,
Dmitry
=
Connected to Asterisk CVS-D2004.07.21.22.00.00-07/23/04-11:25:48 currently
running on linux (pid = 2743)
-- Accepting AUTHENTICATED call from
number there
to be able to talk to him and have him call me.
www.diamondcard.us has affiliate in Kiev and are able to provide Asterisk
solution in Ukraine as well as voip services.
Dmitry
If anyone has any information on it and they are willing to share please
advise.
Thanks,
Jeff
On Wednesday 24 March 2004 19:40, Matthew Marlowe wrote:
I get unable to negotiate codec
GSM and ILBC works.
Dima
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stephen Karrington
Sent: Wednesday, March 24, 2004 10:09 AM
To: Asterisk
more then 10 seconds.
Below is a log of such call.
Its not clear for me why we appear in from_telco context when we are making
a call?
We suppose to be in this context only when we are receiving calls from
telco.
Thanks,
Dmitry.
*CLI == D-Channel on span 4 up
-- Executing Dial(Zap/11-1
information which we are usually getting in CDRs
during the time when the call is still active?
Thanks,
Dmitry
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25 so you may try to use older code from CVS
before this date.
Jeremy saying the latest version approach is fine, but its not working for
me :(.
Dmitry
Can any of your experts out there help please, thanks?
TC
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system
have incorrect codec settings. What is set on client doesn't match oh323
codec settings.
Dmitry
Thanks.
Ehsanul Karim
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Hello!
I am considering mass deployment of Budgetones 102. According to their
website, remote provisioning (configuration via TFTP) is possible.
Anyone has experience with this? Is this really working?
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On Monday 30 January 2006 13:03, Phil Blundell wrote:
Personally I'd be a bit wary of mass Budgetone deployment for other
reasons, but the remote configuration stuff shouldn't be a problem.
What reasons do you mean?
Grandstream use basically the same configuration file system for the
On Monday 30 January 2006 21:48, [EMAIL PROTECTED] wrote:
On Mon, 30 Jan 2006, Dmitry Ivanov wrote:
I have created dynamic CGI-like TFTP server so I will create
config files on-the-fly. Now we use this system (dynamic tftp
server and Perl CGI script) for country-wide Sipura 3000
On Wednesday 01 February 2006 14:12, Cosmin Prund wrote:
As the subject line says: Is PING a good indicator of network
latency? If not, how can I measure latency?
Only if ICMP Echo has the same Class of Service (DSCP, 802.1p,
priority/class in routers/shapers etc.) as VoIP traffic across the
Good day!
Is is possible to change dialtone (and other tones as well) in BT-102?
pgpH067L1PT2h.pgp
Description: PGP signature
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On Tuesday 07 March 2006 15:49, Lee Archer wrote:
Download the IP Phone Custom Ringtones Generation Tool
Unzip and read the readme
Ringtone != dialtone.
pgpEye3ebfT7t.pgp
Description: PGP signature
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channels to make sure the
call went over the e1 channels.
i'm still testing it, it seems to be working great right now only
error i got was a FCS BAD or somthing like that once.
My motherboard is a SuperMicro P4SCI just for your information.
br,
dmitry
Down state
pb01*CLI pri show span 1Primary D-channel:
16Status: Provisioned, Down, ActiveSwitchtype:
EuroISDNType: CPEWindow Length: 0/7Sentrej: 0SolicitFbit:
0Retrans: 0Busy: 0Overlap Dial: 0
Can anybody help me?
br,
dmitry
Dmitry ZhukovskiSystem developerComX
Networks A/SNaverland 31, 2DK
Hi again,
Well - I didn't see beta8a-2.3.3 in custom dir. Will try.
Also I tried to contact Sangoma - they are very fast to answer but main
problem is time difference - it's 6 hours between Canada and Europe.
Br,
dmitry
Dmitry Zhukovski
System developer
ComX Networks A/S
Naverland 31, 2
Run asterisk -vcf for test purpose.
Safe_asterisk is script which runs asterisk - so you wont get any messages on
screen
Br,
dmitry
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Tutu Lord
Sendt: 12 May 2005 09:58
Til: asterisk-users
) ] [Zap/1-1]
And NO MESSAGES on asterisk2!
Looking forward to get any help!
Br,
dmitry
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to contact Sangoma.
Br,
Dmitry
Fra: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Dmitry
ZhukovskiSendt: 20 May 2005 16:12Til:
[EMAIL PROTECTED]Emne: [Asterisk-Users]
ZAP/DTMF
Hi all!
I have got strange problem with DTMF. This
is my test setup
Phone1(#101
Hi
I think you should have a look at the end of line - you are missing :-)
Br,
dmirty
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Ronald
Sendt: 26 May 2005 11:47
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne:
Hallo!
Finally we have E1 PRI connected to our Asterisk box. Now I have one
question.
My internal extensions (_XXX) are SIP phones connected to Asterisk. Our
telco routes some public numbers (_71602XX and others) to our Asterisk
via E1. Some internal extensions can be reached from outside
On Tuesday 28 March 2006 15:40, Tomislav Vojvodic wrote:
It seems it's 'normal' behaviour since I heard exactly the same thing
happening in Croatia. If caller id is set to some number, telco
overrides it to first caller id.. (even if that number belongs to
your block (right?))
No. It sets
On Tuesday 28 March 2006 16:33, Tomislav Vojvodic wrote:
Is that what you were asking?
My question is: how can I set specific caller id for outgoing PRI calls?
pgpqCyPf2WORz.pgp
Description: PGP signature
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Hallo!
Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102.
Looks like none of them works with Mac mini G4...
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On Tuesday 18 April 2006 18:26, The VoIP Connection wrote:
The switch in the Budgetone is 10Base-T. If the PC NIC cannot
auto-detect or otherwise handle that, it will be a problem.
Yes! Looks like Mac mini cannot handle 10HD :) This is what I see when
BT-102 is connected to Alcatel Omnistack
Hi!
Our telco routes multiple numbers through PRI to our Asterisk. Not all
of these numbers are in use. I have noticed recently that someone keeps
calling unused phone number from outside world. I called them and asked
why do they call dead number. The person on the far end explained that
she
On Tuesday 18 April 2006 14:58, Christian Gröger wrote:
Hi,
I am using Asterisk with misdn connected to an ISDN Line, so I have
several numbers I can use...
I know that I can use misdn like this in my extensions.conf:
exten = _0.,1,Dial(mISDN/1/${EXTEN:1})
But how can I use another
On Thursday 20 April 2006 19:22, Kevin P. Fleming wrote:
A better solution is to set the PRI hangup cause before dropping the
incoming call; if you set the hangup cause to 'number not assigned'
then your telco's switch will play its normal intercept message to
the caller.
Thank you! This
On Tuesday 25 April 2006 00:57, Juan Salas wrote:
Hello.
Has anybody knows how run two asterisk process
in one hardware? (each one with its own configuration?)
It is possible.
1) Use different UDP ports for SIP/IAX/RTP
2) Use different log files and astdb files
But most users do not need
On Wednesday 26 April 2006 07:52, Jason Frisch wrote:
How about using time announments? I list of these
for each country would be great!
I have some test numbers on my switch in Latvia:
+371 7160201 -- echo
+371 7160202 -- music :)
+371 7160203 -- time
Do you mean something like this?
On Wednesday 26 April 2006 11:48, Dmitry Ivanov wrote:
On Wednesday 26 April 2006 07:52, Jason Frisch wrote:
How about using time announments? I list of these
for each country would be great!
I have some test numbers on my switch in Latvia:
+371 7160201 -- echo
+371 7160202 -- music
On Friday 28 April 2006 11:35, Ralf Schlatterbeck wrote:
I don't see the call at all in asterisk.
Maybe your telco does not route these calls with incomplete number to
you?
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On Wednesday 03 May 2006 11:08, Tomislav Parčina wrote:
Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that
cost around 100USD, and those phones are one of options.
Can anybody suggest anything else that costs around 100USD?
We have 10 Grandstream Budgetones 101 102 here in
Hello!
First problem with 1.2-beta2.
All I hear during Echo() is noise. No matter which codec selected.
However, when using ulaw noise sounds better than g723 :)
My equipment is Sipura SPA-3000. Works fine with 1.0.9 amd 1.0.7.
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Hello!
We consider purchasing Digium Wildcard for E1 connectivity. Wildcards
are pretty expensive pieces of silicon for small shop like ours. And we
have no previous experience with E1 communications.
What Wildcard do we need? How can we estimate our needs? How many
clients (approx.) can
On Tuesday 08 November 2005 11:34, Hugh Jackman wrote:
Hi,
Digium have Wildcard for FXO/FXS connections (i.e., telephone lines)
and E1/T1 cards such as the TE110p. There're a few things you might
want to consider:
1) TE110p is much more expensive
2) it is too much for a small shop.
On Tuesday 15 November 2005 06:26, [EMAIL PROTECTED] wrote:
Hi. I'm setting up an Asterisk hobby box for me to play around with.
Is it possible to use a regular 56k modem and a regular home phone
for it?
Yes, but forget G.711.
BTW, some SIP-phones have built-in modem :)
On Thursday 17 November 2005 12:32, Anton Krall wrote:
Drumroll
TADA!!
Already compiling it :)
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On Thursday 17 November 2005 12:58, Dmitry Ivanov wrote:
On Thursday 17 November 2005 12:32, Anton Krall wrote:
Drumroll
TADA!!
Already compiling it :)
Unlike beta2, it works for me. I hear no noise during echo test.
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-dial extension number
-If there is no answer at this extension start voicemail.
Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already answered the call?
Thanks,Kupchinetsky Dmitry
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to answer
-dial extension number
-If there is no answer at this extension start voicemail.
Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already answered the call?
Thanks,Kupchinetsky Dmitry
___
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to answer
-dial extension number
-If there is no answer at this extension start voicemail.
Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already
answered the call?
Thanks,
Kupchinetsky Dmitry
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point me to technical materials or show electric scheme of such converter. I
believe it should be rather simple.
I'm ready to cooperate to make such solution.
Thanks,
Dmitry.
_
The new MSN 8: advanced junk mail protection and 2 months
point me to technical materials or show electric scheme of such converter. I
believe it should be rather simple.
I'm ready to cooperate to make such solution.
Thanks,
Dmitry.
_
The new MSN 8: advanced junk mail protection and 2 months
.
Dmitry
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Hi everyone!
Does anybody have russian (or/and ukrainian) language sound file
collection for Asterisk? (at least those required for the voicemail
system). Or maybe anyone could point me to the location where to get ones?
TIA
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Hi all!
I have got a bit strange output from iax2 show channels:
Med venlig hilsen
ComX Networks A/S
Dmitry Zhukovski
System developer
ComX Networks A/S
Naverland 31, 2
DK-2600 Glostrup
Denmark
Phone: +45 70 25 74 74
Fax: +45 70 25 73 74
Web: www.comx.dk
Dmitry Zhukovski
Direct
5448 2556 S 99.9 0.3 1439:20 asterisk
1 root 16 0 680 248 216 S 0.0 0.0 0:01.22 init
2 root RT 0 000 S 0.0 0.0 0:00.00 migration/0
Any ideas? Thank you in advance,
Dmitry
Med venlig hilsen
ComX Networks A/S
Dmitry Zhukovski
System developer
Hello!
Is it possible to connect Siemens HiPath 4000 to Asterisk? What
equipment required on Siemens side? I mean IP not E1.
Sorry for asking here. Siemens-related websites use salesperson
language. There is no technical information.
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On Wednesday 21 December 2005 15:11, [EMAIL PROTECTED] wrote:
Is Asterisk able to provide virtuals IPBX ?
I mean one hardware server which handle one IPBX per
enterprise .
Yes, just create separate context for each enterprise.
___
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On Thursday 22 December 2005 07:36, Stefan Reuter wrote:
http://www.voip-info.org/wiki-Asterisk+cmd+Monitor
For Asterisk 1.2:
http://www.voip-info.org/wiki/view/MixMonitor
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Is is possible to select (preferred) codec in dialplan using
extensions.ael? For example, use 711 for extension 6004 (which is not
physical extension) and 729 for anything else?
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On Friday 23 December 2005 10:22, Mauro Zanin wrote:
Hi everybody,
no issues this time. Only stopped to say: Merry Christmas and Happy
New Year.
Yes, Merry Christmas, Happy New Year and Hanukkah :)
Just received nice postcard from Digium :)
___
Hello!
After many troubles, I have received my Wildcard 406. There is a label
on antistatic bag stating that this is 406. The card itself is marked
as 405. Kernel modules shows in dmesg that card is 405.
Is 406 the same as 405 with additional board installed?
On Tuesday 10 January 2006 13:06, Jean-Michel Hiver wrote:
http://lists.digium.com/pipermail/asterisk-users/
May 2016? November 2007? Woot? Some kind of delayed Y2K bug?
Randal Law lives in future.
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http://bugs.digium.com/view.php?id=7451 )
I guess that other voicemail DB storage configuration is known for you.
Regards,
Dmitry V.Furmanov
Hello all,
Newbie here, been searching the net to follow the development and feedback
on the 1.2 release feature with using an odbc storage for storing
the asterisk with voicemail DB storage support.
Regards,
Dmitry V.Furmanov.
Hi Dmitry,
Thanks so much for replying, I had a read of the bug descriptions you pointed to but unfortunately am not a programmer so have a bit of a struggle following the entire discussion. I actually AM using 1.2.9.1. So
://bugs.digium.com/view.php?id=7451 )
4. Rebuild and reinstall the Asterisk
Then asterisk will be be able to react on the ODBC configuration
parameters pooled and poolsize.
Regards,
Dmitry V.Furmanov
Hi Dmitry
Thanks for that. Yes, I had done all of that and have a working Asterisk - MS SQL setup
(feature). Because ODBC was specified as
universal database access engine. So, some database servers possibly
haven't substance like view.
Regards,
Dmitry V.Furmanov
RR пишет:
Hi Dmitry,
just to answer your questions, and telling you what I've done so far,
1) yes, using FreeTDS
2) yes
Hi all,
I have got same problem - bridging between IAX and IAX goes fine without lost
packets. ZAP to IAX - one lag show lost packets. Any ideas and/or solutions?
Best regards,
Dmitry
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Simone
will hear out of range message similar to mobile
networks. Is this possible?
--
Dmitry Ivanov
Network engineer
Telecentrs Riga, Latvia
[EMAIL PROTECTED]
(+371) 7160235
Weather at Riga Intl (EVRA/RIX): Monday 06 November 2006 10:50,29
km/h SSE,-4°C,1004 hPa,Broken clouds at 396 meters;Overcast
clouds
On Monday 06 November 2006 16:41, Matt wrote:
This should work.. please make sure you have qualify=yes on in
your sip.conf file for each of your sip entries.
Now it works. Thank you!
On 11/6/06, Dmitry Ivanov [EMAIL PROTECTED] wrote:
Hello!
I have this in my dialplan:
Dial
Hello!
I run asterisk 1.6.1.0, dahdi 2.1.0.4 with TE122.
I always (and only) have missed interrupt when dahdi disables echo
canceller (ng2 or oslec- no difference).
Dec 29 14:00:54 asterisk kernel: dahdi: Disabled echo canceller because
of tone (rx) on channel 1
Dec 29 14:00:54 asterisk
http://leyvacrystaljd.blog23.com
_
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969--
Hello!
I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11)
There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP
But I (and my users) do bridged calls from dahdi to sip, so in h
extension channel is dahdi , and it doesn't contain rtcp stats.
There is info about
Hello!
Could somebody tell me how to use SHARED function?
I want to get RTCP stats from SIP , but current channel is DAHDI.
How can I get SIP channel?
--
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21.09.2010 18:57, Philipp von Klitzing пишет:
Hi!
Could somebody tell me how to use SHARED function?
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+shared
There are no examples there :-(
I want to get RTCP stats from SIP, but current channel is DAHDI.
How can I
22.09.2010 14:50, Philipp von Klitzing пишет:
Hi!
I see. I want to use SHARED function!
Do you have example how to
to export them to the local call leg/channel ?
Have you considered using Google (or your favourite search engine)?
Shure, I searched and find nothing.
The
22.09.2010 15:12, Andrea Cristofanini пишет:
Could you, please, give me link ? :-)
Google is not difficult to use... BTW
http://www.voip-info.org/wiki/view/Asterisk+func+shared
There is no example here!
I already wrote about this...
--
22.09.2010 16:08, Philipp von Klitzing пишет:
Hi Dmitry!
Hello!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer in
the archive of this list you
23.09.2010 16:06, Philipp von Klitzing пишет:
Hi!
And the third hit in my google result is this:
http://lists.digium.com/pipermail/asterisk-users/2009-July/235288.html
Since I mentioned in my previous message that you will find the answer
in the archive of this list you could have
://lists.xensource.com/archives/html/xen-users/2006-04/msg00032.html
To sum things up, I think virtualization is a good idea, especially when
you have beefy servers (I do it too). So a green light from me!
All the best,
D.
--
Dmitry Nedospasov dmi...@nedos.net -- Twitter: @nedos
Web: http://nedos.net
Oops forgot one thing,
On Sun, Sep 26, 2010 at 10:45:38PM +0200, Dmitry Nedospasov wrote:
On Sun, Sep 26, 2010 at 01:48:40PM -0400, bruce bruce wrote:
I am stack between two identical systems (2U Twin2, 4 nodes, SuperMicro)
servers that have the same exact specs except for HDDs
23.09.2010 17:27, Philipp von Klitzing пишет:
Hi!
There are 2 things I can't understand
- 1. how can I know channel name?
${CHANNEL}
Thank you!
Really, I get SIP channel name in macro (I thought I'll still have DAHDI
there...)
2. where should I call this SHARED function?
27.09.2010 09:26, Dmitry Melekhov пишет:
Since I have not done this with 1.6 or 1.8: See if you can get the RTCP
data without using CHANNEL(), and instead use the individual
xxxBRDIGED
RTCP channel variables as illustrated on the Wiki. Your SIP channel is
the 2nd channel (= the bridged one
27.09.2010 16:25, Philipp von Klitzing пишет:
Hi!
Well, only problem I see, is to how pass channel name from macro to h
extension...
SHARED() or CDR(userfield)
Philipp
Looks like I still don't understand how SHARED works :-(
Let's say, I dial my softphone:
28.09.2010 15:35, Philipp von Klitzing ?:
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Hello!
Thank you!
I can pass this constant , but I need RTCP stats
And this
28.09.2010 16:19, Dmitry Melekhov ?:
btw, about bridged variables- they are really what I need.
Looks like there is bug in asterisk- if call is dropped from dahdi side-
there is no info in these variables.
I think I have to fill bug.
Thank you!
I got what I want :-)
Thank you again
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