Hi All,
I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP.
On incoming calls from Avaya asterisk complains of 'unsupported crypto
parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not
acceptable here'
Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto
you see any harm in this solution? Can you suggest me a better solution?
I'll appreciate your responses.
Thanks,
--Satish Barot
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Hi All,
I have been working on a project where I need to record a call in Asterisk
and then split the recording into multiple audio files based on a presence
of particular sound (i.e. beep) in a recording.
I know this is out of scope for Asterisk but I wanted to benefit from
someone else's
. It is possible with asterisk
or not.
thanks in advance.
Regards
Akhilesh
Chanspy with w option
- w - Enable whisper mode, so the spying channel can talk to the
spied-on channel.
https://wiki.asterisk.org/wiki/display/AST/Application_ChanSpy
--Satish Barot
satish4aster...@gmail.com
://wiki.asterisk.org/wiki/display/AST/ManagerAction_Redirect)
and
Bridge (https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Bridge)
--Satish Barot
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is pressed.
--Satish Barot
Ahmedabad, India.
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Yes you can. Check the 'context' parameter in queues.conf. When caller
presses a single digit extension while waiting in a queue, (s)he'll be
taken out of queue to this context. Then you can send caller to different
queue from this context.
--Satish Barot
Ahmedabad, India.
+919978599700
On Wed
:)
--Satish Barot
Ahmedabad, India
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asterisk
On Thu, Jul 4, 2013 at 5:36 PM, Administrator TOOTAI ad...@tootai.netwrote:
Le 04/07/2013 07:29, Satish Barot a écrit :
[...]
Already tested, I tried again as the option passed to queue was
changed (n option)
Logs:
-- Started music on hold, class 'default', on SIP/gw
}.wav)
exten =
_4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
exten = _4X.,n,Dial(SIP/${EXTEN},30)
exten = _4X.,n,Hangup
Regards
On 4 Jul 2013 11:18, Satish Barot satish4aster...@gmail.com wrote:
On Thu
--Satish Barot
Ahmedabad, India
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On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI ad...@tootai.netwrote:
Hi Satish
Le 03/07/2013 09:15, Satish Barot a écrit :
On Tue, Jul 2, 2013 at 10:57 PM, Administrator TOOTAI
ad...@tootai.netmailto:
ad...@tootai.net wrote:
Hi all,
I have to questions about queues
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI ad...@tootai.netwrote:
Le 03/07/2013 15:07, Satish Barot a écrit :
[...]
Then you should add Local channel as a queue member and dial your SIP
member from Local channel context. A little hint here. Suppose you have a
support queue
:17, Satish Barot satish4aster...@gmail.com wrote:
And yes if you want to use System application in your dialplan then have
System in your h extension
System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav
/PathToMp3FileToBE Stored/filename.mp3)
On Tue, Jun 11, 2013 at 10:38 AM
any issue.
call-limit I think is deprecated in 1.8.
--Satish Barot
Ahmedabad, India
On Sat, Jun 22, 2013 at 2:41 PM, Shanavaz E A shanava...@yahoo.com wrote:
Hi,
I use asterisk 1.8.
My issue is : I have the same SIP members added to two queues. I use
realtime configuration and has set
data
through webservice call ().
do you want to use C or PHP?
-Thorsten-
Hi Thorsten
Normally I use 'PHPAGI' in my Asterisk applications but as I said want to
explore C as an option.
Thanks,
--Satish Barot
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On Thu, Jun 20, 2013 at 10:54 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Thu, 20 Jun 2013, Satish Barot wrote:
Would you mind sharing a sample of your pthread-ed C AGI? This will help
someone like me who has written AGI in Perl/PHP and now exploring C AGI.
The source code
(- CALL DIDN'T GET ANSWERED IN FIRST LEG -)
--Satish Barot
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On Mon, Jun 17, 2013 at 7:22 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 17 Jun 2013, Thorsten Göllner wrote:
does anyone have experience with Asterisk-AGI-Scripts in PHP while using
pthreads in PHP? Are there any limitations or problems known?
I've written 'pthread-ed' AGIs
should have something like
*MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav)
in your dialplan.
Hope this helps.
--Satish Barot
Ahmedabad, India
On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi Satish,
I tried with sox
And yes if you want to use System application in your dialplan then have
System in your h extension
System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav
/PathToMp3FileToBE Stored/filename.mp3)
On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot satish4aster...@gmail.comwrote:
Hi
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
promiscredir= yes in sip.conf should help you achieve your requirement.
I haven't been able to get that to work in a similar situation, except we
On 5/9/13, Satish Barot satish4aster...@gmail.com wrote:
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
promiscredir= yes in sip.conf should help you achieve your requirement.
I haven't been able to get
= _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS})
However, this does not work,
Is there a way to send the 302 response to the VoIP provider ?
Thanks.
Hans
promiscredir= yes in sip.conf should help you achieve your requirement.
--Satish Barot
Ahmedabad, India
(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
Further your readsql should be like this.
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${ARG1})}'
You should have ${ARG1} instead of ${CONF_ID}
Hope this helps
--Satish
On Fri, Apr 19, 2013 at 5:59 PM, Satish Barot satish4aster...@gmail.comwrote:
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.netwrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users
@extensions)
[extensions]
exten = _X.,1,Dial(SIP/${EXTEN})
same = n,Execif($[${DIALSTATUS}=BUSY]?Answer():)
I couldn't test the code and has obvious side effects on CDR.
--Satish Barot
Ahmedabad, India.
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not exist.
Thoughts ?
Regards
You can achieve the same functionality using IF function. Something like,
...
same = n,Set(foo=${IF($[ ${DB(family/key)} =
]?defaultval:${DB(family/key)})})
--Satish Barot
Ahmedabad, India
On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Monday 08 April 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session
,UnpauseQueueMember(,Local/${ARG2}@to-${myQueue})
same = n,UnpauseQueueMember(,Local/${ARG3}@to-${myQueue})
same = n,UnpauseQueueMember(,Local/${ARG4}@to-${myQueue})
Hope this helps.
--Satish Barot
Ahmedabad, India
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that your
headphones work properly every time you start the softphone? This has
happened to me in past.
--Satish Barot
Ahmedabad, India.
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value for QUEUE_PRIO varibale in Server X dialplan for calls
coming from server A.
If you do not wish to drop the calls when no agent is available to take the
call(either she is busy on call or in pause mode), set joinempty = yes
and leavewhenempty
=no in queues.conf
--Satish Barot
Ahmedabad, India
in advance
Ding Peng
https://wiki.asterisk.org/wiki/display/AST/Home is the best place to
start off with such stuffs.
--Satish Barot
Ahmedabad, India
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New
/05/how-to-install-asterisk-11-on-centos-6/
--Satish Barot
Ahmedabad,India
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interval. Use first leg of a call to
Chanspy with Bargein mode on your channel and second leg of a call to
Playback a message.
--Satish Barot
Ahmedabad,India.
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= n,ExecIf($[${QUEUESTATUS} = CONTINUE]?Set(HNGPPARTY=AGENT):)
... ...
exten = h,1,set(CDR(userfield)=${HNGPPARTY})
Note that if nobody answers a call then a variable HNGPPARTY would be empty.
--Satish Barot
Ahmedabad,Gujarat,India
On Fri, Jan 11, 2013 at 10:29 AM, Satish Barot satish4aster...@gmail.comwrote:
On Thu, Jan 10, 2013 at 7:53 PM, RSCL Mumbai rscl.mum...@gmail.comwrote:
Hello,
Can asteriskCDR logs tell me if a call was disconnected by the caller
or the Agent ?
My call flow is as follows:
Caller Dials
HI Andrew,
Show your queuecontrol context. You should have extension s with priority
1 in this context.
--Satish Barot
On Mon, Jan 7, 2013 at 12:08 PM, Andrew White and...@computersforall.com.au
wrote:
Hi Satish,
** **
Thanks for your response – sorry on the slow reply
/AST/Application_Record
--Satish Barot
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On Mon, Dec 31, 2012 at 3:28 PM, Satish Barot satish4aster...@gmail.comwrote:
On Mon, Dec 31, 2012 at 3:12 PM, Vinod Nadiadwala thinw...@gmail.comwrote:
Hi,
I am new to asterisk, i want to know that is it possible to use asterisk
for build voice recording system.
Scenario :
ISDN PRI line
the link for more information, Arguments are passed to subroutine using
^ as a delimiter.
--Satish Barot
** **
*From:* Andrew White
*Sent:* Wednesday, 19 December 2012 5:58 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Dialplan
://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
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application 'originate' in place of callfiles. I
normally prefer local channels in Callfiles or Originate so that I can have
better call control through dialplan.
--Satish Barot
On Mon, Dec 3, 2012 at 3:08 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
How I can acheive the following
I put ${CHANNEL(dahdi_span)} to know the span and ${CHANNEL(dahdi_channel)}
for actual channel number in incoming context of PRI.
For outbound I normally use M flag in Dial() to call a macro and check the
above variables in that macro.
--Satish Barot
On Tue, Nov 6, 2012 at 7:02 PM, Amit Patkar
(**num)} = 2024324321]?other,${**thisexten}:)
--Satish Barot
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a question!
Check a function IAXVAR.
I think Asterisk version matters for it.
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in sales Queue and then check the
value for variable.
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Hi Herve,
Asterisk is legal in India and using it for Fax shouldn't create any issues
as far as legality is concerned.
Look at following link to get some idea on VoIP regulation in India.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fslopar.html#wp1114625
--Satish Barot
Hi Akhilesh,
Probably this link would give you some idea on ASR. With the help of it,
add some logic in dialplan to develop an application of your choice.
(Courtesy Lefteris Zafiris)
Goto https://github.com/zaf/asterisk-speech-recog/ and read README
--Satish Barot
On Thu, Jul 5, 2012 at 12:46
are free, Generate a callfile OR use AMI to
call the caller who has requested a callback.
(6)Once call is answered, send him to Queue application with 'position'
parameter set to the value of 'QUEUEPOSITION' of caller from database.
--Satish Barot
On Thu, May 31, 2012 at 9:18 PM, equis software
Yes of course you can use local channel with AddQueueMember().
--Satish Barot
On Wed, Apr 11, 2012 at 1:22 PM, Olivier oza_4...@yahoo.fr wrote:
2012/4/11, Satish Barot satish4aster...@gmail.com:
I would implement it in a different way.
As you seem to be a seasoned player just a hint here
22, 2012 at 10:33 AM, Satish Barot
satish4aster...@gmail.comwrote:
Make your user wait in a *Meetme* and then call your destination number
through AMI and once he answers, place him in the same *Meetme*.
e.g. Assuming your destination is SIP extension, have something like...
Action
: {your_meetme_number_here}
Hope this helps.
--Satish Barot
On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar jayesh.v...@gmail.comwrote:
Hello All,
I need to know a way of connecting an Answered call in Asterisk to another
call which was triggered by an AMI. I have a scenario as follows:
1) User dials
as a Queue member and have your local channel dial the cellphone or
Landline number.
See the 'Using Local Channels' section on a link
http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html for more
information. (Courtesy:Leif Madsen, Jim Van Meggelen, and Russell Bryant)
--Satish Barot
On Wed
Hi Virendra,
I should have said, you can *set the callerid to one of the numbers
allocated by them* for PRI, * and not to any other number*.
Enjoy.
--Satish Barot
On Tue, Feb 14, 2012 at 1:31 PM, virendra bhati virbh...@gmail.com wrote:
Satish,
As if I know, PRI provider give you PRI number
Indian Telcos do allow setting callerid on PRI line and you can set the
callerid to one of the numbers allocated by them for PRI.
--Satish Barot
On Mon, Feb 13, 2012 at 6:49 PM, Ast Coder asteriskcod...@gmail.com wrote:
India TRAI rules doesn't allow for CLID setting. They are backwards
/asterisk/mp3/$MP3
/usr/bin/lame ${WAV} ${MP3DEST} --silent -b 16 -s 9.6 -m m --bitwidth 8
--lowpass 9.6 --resample 8 --lowpass-width 1
--SATISH BAROT
Ahmedabad,India.
On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib fkha...@iconnecths.comwrote:
Hello Guys,
I am trying to convert files
,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati virbh...@gmail.com wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user
,
--SATISH BAROT
On Wed, Dec 28, 2011 at 3:02 PM, virendra bhati virbh...@gmail.com wrote:
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling
Check 'retry' in queues.conf
[SATISH]
Mumbai, India.
On Sun, Jul 10, 2011 at 4:34 PM, Florent THOMAS mailingl...@tdeo.fr wrote:
Hy,
I'm currently working with one queue and whatever I change in the config,
it stills a gap of 6 seconds during which no agents are ringing for this
queue.
Is
What do you mean by ring time out?
See the 'timeout' in Queue application. keep it blank if you just want to
keep your callers in queue for infinite time.
Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule[,position])
[SATISH]
Mumbai, India
On Fri, Jul 8,
Would this be of any help to you?
http://lists.digium.com/pipermail/asterisk-users/2011-June/263339.html
[SATISH]
Mumbai, India.
On Mon, Jun 27, 2011 at 7:14 AM, Rafael dos Santos Saraiva
rafaels...@gmail.com wrote:
I am referring to 3-way conference
Att,
Rafael Saraiva
2011/6/26
Wasn't that helpful?
http://lists.digium.com/pipermail/asterisk-users/2011-June/264082.html
Use GotoIfTime in agi of your choice with condition part being calculated
dynamically as per your requirement.
But I really don't see any usefulness of AGI if your working hours are fixed
i.e. Mon - Thu,
Would it create any problem for Asteisk, if we install Windows as a VM on a
system that has CentOS running Asterisk as the base?
System also has a PRI card.
TYIA,
[SATISH]
Mumbai, India.
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then it will not be
more than a part sharing from system resources depends on VM configuration
and processing load.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Satish Barot
*Sent:* Friday, June 24, 2011 12:38 PM
*To:* Asterisk Users
Hope following will help you get some idea.
[default]
exten = _45789XX,1,Set(VMNO=${EXTEN:-2})
same = n,GotoIfTime(9:00-19:00,sun-thu,*,*?:NON-WORKING-HRS,s,1)
...
...
[NON-WORKING-HRS]
exten = s,1,Playback(non-working-hrs)
exten = s,n,VoiceMail(50${VMNO})
exten = s,n,Hangup
[SATISH]
Mumbai,
Hi all,
I will really appreciate if you can spend some time to share your experience
or point me in right direction.
I have been told to prepare a single box Asterisk system (No Distributed
architecture) for following features.
-Asterisk 1.8
-300 SIP extensions (sip.conf)
-8 port PRI card (E1)
Check the option of 'd' in Dial().
d: Allow the calling user to dial a 1 digit extension while waiting for a
call to be answered. Exit to that extension if it exists in the current
context, or the context defined in the ${EXITCONTEXT} variable,if it exists.
[SATISH]
On Wed, Jun 15, 2011 at
ringinuse=yes will send your call to Agent only if her phone state is in 'In
use' or 'Not in use' BUT not when it is in
'ringing'.
So probably you can not achieve what you want with the current Queue()
implementation in Asterisk.
[SATISH]
On Mon, Jun 13, 2011 at 4:14 PM, Deka, Rajib IN MAA SL
database for the sip users. For my config I am using
OpenSIPS as the register and proxy. Asterisk is only used for voicemail and
ACD/Hunt groups.
On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot
satish4aster...@gmail.comwrote:
Provide the entry for Agent SIP/9013XX9XX8 along with parameters
Provide the entry for Agent SIP/9013XX9XX8 along with parameters
'callcounter' and 'qualify' from sip.conf.
Also provide CLI outputs of 'core show channels',sip show peers' and 'queue
show' when...
(1)First caller enters the Queue
(2)First caller gets connected with Agent
(3)First caller gets
Set queue_log = no in logger.conf. By default it is set to 'yes'.
[SATISH]
On Wed, Jun 8, 2011 at 12:30 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
I have configured extconfig.conf to save queue log into my MySQL-DB.
I notice however that there is still logging too in
Give it a shot and check! :)
Yes you will have your Queue log records in table.
[SATISH]
On Wed, Jun 8, 2011 at 12:46 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 06/08/2011 09:10 AM, Satish Barot wrote:
Set queue_log = no in logger.conf. By default it is set to 'yes'.
[SATISH
I hope my understanding is not wrong!
(1) DAHDI/i2/25/XXX, is not a valid format for Dial. Rather it
should be DAHDI/i2/XXX and it would use a channel from span 2
(/etc/dahdi/system.conf) for outgoing call.
(2) To dial from channel 25 , use DAHDI/25/XXX
[SATISH]
On
How do you want to map callerid with your extensions? Do you have any DB
table for such a mapping?
[SATISH]
On Tue, Jun 7, 2011 at 2:29 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was
)=${outgoing_ident}), will
always set callerid to 044578900 for Extensions 100,200,300; 044578901 for
101,201,301 and so on .
[SATISH]
On Tue, Jun 7, 2011 at 5:11 PM, mahesh katta maheshka...@flexydial.comwrote:
Sir,
I have MYsql database in myserver.
On Tue, Jun 7, 2011 at 4:57 PM, Satish
I use following for MySQL...
CREATE TABLE queue_log(
id int(11) NOT NULL auto_increment,
time datetime not null,
queuename VARCHAR(50),
agent VARCHAR(50),
callid varchar(32),
event VARCHAR(100),
data1 VARCHAR(100),
data2 VARCHAR(100),
data3 VARCHAR(100),
data4 VARCHAR(100),
data5 VARCHAR(100),
It would be a great help to others(including me) if those using 1.8.X can
provide some details on hardware configurations,features they have
implemented on it and some sort of load testing results.
Thanks,
[SATISH]
On Mon, Jun 6, 2011 at 6:28 AM, Sherwood McGowan sherwood.mcgo...@gmail.com
If 1.8 doesn't panic for subset of PBX features for someone, you can not say
it is stable. You should also look at other
features and how they work with 1.8.
I didn't say 1.4 or 1.6 have no bugs or issues. When there were 1.4 or 1.6.0
branches, they did have bugs. But since people
started
See this link for release date...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
[SATISH]
On Thu, Jun 2, 2011 at 1:09 PM, Nikhil d.nik...@cem-solutions.net wrote:
I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt
find this release from asterisk
Nikhil,
This is how I would implement '3 way conference' in Asterisk with the help
of dynamic features.
Assume 3 SIP friends 1110, and 1112 in sip.conf. For 1110 in sip.conf,
context=test3way
Add following in applicationmap section of features.conf
[applicationmap]
3way-start =
So many new features have been added in 1.8.
Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8
Nope, Asterisk 1.8 is not stable enough yet.
[SATISH]
On Thu, Jun 2, 2011 at 6:33 PM, Gopal krishnan
gopalakrishnan...@gmail.comwrote:
1.8 is stable when compared to 1.6, also in
Paul,
With due respect to Digium work, are there no issues with Asterisk 1.8?
https://issues.asterisk.org/view_all_bug_page.php
[SATISH]
On Thu, Jun 2, 2011 at 9:21 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 06/02/2011 10:29 AM, Eric Wieling wrote:
-Original Message-
Use Asterisk Application 'System()' in h extension to
create callfile which will handle your callback.
You can also try for 'Originate()' application.
[SATISH]
2011/6/3 Antonio Modesto mode...@isimples.com.br
Good afternoon,
I'm trying to write a simple callback context, but i need to
Warren,
A good example given.
Just suggest to use 'Move' instead of 'Copy' for placing callfile in
outgoing folder.
A J Stiles has explained it in a better way in one of his replies.
http://lists.digium.com/pipermail/asterisk-users/2011-May/262929.html
[SATISH]
On Fri, Jun 3, 2011 at 1:16 AM,
Hi Everybody,
Don't know why this DBdeltree error appears on Asterisk CLI.Good part is, it
does remove family entry from AstDB.
Sample Dialplan
exten = 1212,1,Noop()
same = n,Set(TEST=1234)
same = n,Set(DB(${TEST}/TESTSTART)=${STRFTIME(${EPOCH},,%Y-%m-%d
%H:%M:%S)})
same =
While playing with DB function in Dialplan, I have added some garbage in
AstDB. These are some family names with space in them.
See this,
demo*CLI database show
/18-05-2011 00:00:0052011175221575/TESTDATE: 2011-05-14 21:33:46
/18-05-2011 00:00:0052011175221575/TEST1 : 410
/18-05-2011
.
--
Regards,
Chandrakant Solanki
On Mon, May 30, 2011 at 2:53 PM, Satish Barot
satish4aster...@gmail.comwrote:
While playing with DB function in Dialplan, I have added some garbage in
AstDB. These are some family names with space in them.
See this,
demo*CLI database show
/18-05-2011
If you go for 1.8,Don't read from
http://www.asteriskguru.com/tutorials/queues.html. It is bit backdated
information. Rather I would suggest you to check
http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html.
Queue members are considered INVALID, if their device status is Invalid.
This
If you don't like callfiles, another option is AMI. Check the sample code
from
http://tycoontalk.freelancer.com/php-forum/156207-click-to-call-using-php.html,
do some changes as per your requirements.
I would love to use callfiles as it gives more flexibility(as per my
understanding) compared to
I was looking for MySQL table structures for ARA (Asterisk 1.8.X).
I found one for SIP friends on,
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure
But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it
contains 'call-limit' which is deprecated in
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