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 parameters
quot;D"]?NoOP(D received):HangUp())
same => n,MixMonitor(audiofile2)
... ...
Do you see any harm in this solution? Can you suggest me a better solution?
I'll appreciate your responses.
Thanks,
--Satish Barot
--
_
-- Band
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 experie
s 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
-
wiki/display/AST/ManagerAction_CoreShowChannels),
Redirect (https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Redirect)
and
Bridge (https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Bridge)
--Satish Barot
Ahmedabad, India.
--
__
macro nway_start, You can safely assume that only
0 is pressed.
--Satish Barot
Ahmedabad, India.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar eve
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.
+919978
IXMONITOR_FILENAME}.wav,b,/root/flac.sh
> ${MIXMONITOR_FILENAME}.wav)
> exten =>
> _4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
> exten => _4X.,n,Dial(SIP/${EXTEN},30)
> exten => _4X.,n,Hangup
&g
On Thu, Jul 4, 2013 at 5:36 PM, Administrator TOOTAI wrote:
> 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:
>>
>>
${CALLERID(num)} should give you only number and not technology i.e. 41712.
Give this a shot,
exten => _417XX,n,Noop(CALLERIDNUM=${CALLERID(num)})
exten => _417XX,n,GotoIf($[$["${CALLERID(num)}" > "41799"] |
$["${CALLERID(num)}" < "41700"]]?
rmat..
> On 11 Jun 2013 11:17, "Satish Barot" 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
>> /PathToMp3File
On Wed, Jul 3, 2013 at 7:40 PM, Administrator TOOTAI wrote:
> 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. Suppo
On Wed, Jul 3, 2013 at 2:37 PM, Administrator TOOTAI wrote:
> 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.net>> wrote:
>>
>> Hi all,
>>
>>
=8 in
> queue conf, how to tell asterisk to retry each 20 seconds playing MOH to
> the caller?
>
> Thanks for any hint
>
> --
> Daniel
>
>
--Satish Barot
Ahmedabad, India
--
_
-- Bandwidth and Colocation Prov
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 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
r while passing/processing some data
>> through webservice call ().
>>
>>
> do you want to use C or PHP?
>
> -Thorsten-
>
Hi Thorsten
Normally I use 'PHPAGI' in my Asterisk applic
On Thu, Jun 20, 2013 at 10:54 PM, Steve Edwards
wrote:
> 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 sour
On Mon, Jun 17, 2013 at 7:22 PM, Steve Edwards wrote:
> 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 in C.
>
> The on
n => outbound1,n,Set(recipient=${recipient})
>
> exten => outbound1,n,Dial(SIP/${recipient}@originateChannel)
>
> ****
>
> Anyone have an idea how to fix this?
>
>
> --
>
You need a special extension 'failed' in a context originateDialProcessor
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 wrote:
> Hi Gopamkrishnan,
>
>
o
my script.
*
*You 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
On 5/9/13, Satish Barot wrote:
> On 5/9/13, Carlos Alvarez wrote:
>> On Tue, May 7, 2013 at 10:05 PM, Satish Barot
>> wrote:
>>
>>>
>>>
>>> promiscredir= yes in sip.conf should help you achieve your requirement.
>>>
>>
>> I ha
On 5/9/13, Carlos Alvarez wrote:
> On Tue, May 7, 2013 at 10:05 PM, Satish Barot
> wrote:
>
>>
>>
>> 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 w
; internal_devices
> exten => _X.,1,Verbose(1,${CALLERID(num)} tries call forward to ${EXTEN} for
> device ${CALLERID(rdnis)})
> exten => _X.,n,Transfer(${EXT_TRUNK}/${EXTEN})
> exten => _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS})
>
> However, this does not work,
>
> I
On Fri, Apr 19, 2013 at 5:59 PM, Satish Barot wrote:
>
>
>
> On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins wrote:
>
>> All,
>>
>> Thank you in advance for any help.
>>
>> I have a customer in need of a conferencing system. A requirement is fo
xten=>_XX,n,ODBCFinish()
exten=>_XX,n,Goto(cleanup,1)
exten=>cleanup,1,Verbose(1,Finish up)
same=>n,Verbose(1,PIN not found)
same=>n,ODBCFinish(${ODBC_ID})
same=>n,playback(conf-invalidpin)
same=>n,Goto(rooms,${CONF_ID}1)
exten=>good_exten,1,Verbose(1,The PIN is ava
gly way to achieve this!
exten => 100,1,Dial(Local/101@extensions&Local/102@extensions)
[extensions]
exten => _X.,1,Dial(SIP/${EXTEN})
same => n,Execif($["${DIALSTATUS}"="BUSY"]?Answer():)
I couldn't test the code and has obvious side effects on CDR.
--Satish
/key
> value does not exist.
>
> Thoughts ?
>
> Regards
>
You can achieve the same functionality using IF function. Something like,
...
same => n,Set(foo=${IF($[ "${DB(family/key)}" =
""]?de
On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles wrote:
> On Monday 08 April 2013, Thomas Perron wrote:
> > I am trying to make sure my DID and SIP account details are working
> > properly and engaging the extensions.conf and dial plan.
> >
> > I have a successful SIP session registered:
> >
> > Connect
ocal/${myExten}@to-${myQueue}))
;;same => n,Set(__${myQueue}STATUS=myTIMEOUT)
same => n,Set(SHARED(${myQueue}STATUS,${PARENTCHANNEL})=myTIMEOUT)
;;same => n,NoOp(Value of my variable is ${${myQueue}STATUS}) ; here I get
correct value
; ${myQueue}STATUS is set here through SHARED FUNCTION
ttp://lists.digium.com/mailman/listinfo/asterisk-users>
>
Silly guess, If there is no then NAT did you check that your
headphones work properly every time you start the softphone? This has
happened to me in past.
--Satish Barot
Ahmedabad, India.
--
_
gt;
> please guide me
>
> Regards
> Akhilesh
>
>
Set higher value for QUEUE_PRIO varibale in Server X dialplan for calls
coming from server A.
If you do not wish to drop the calls when no agent is available to take the
call(either she is busy on call or in pause mode), set joinempty
;
> Thanks in advance
>
> Ding Peng
>
>
> https://wiki.asterisk.org/wiki/display/AST/Home is the best place to
start off with such stuffs.
--Satish Barot
Ahmedabad, India
--
_
-- Bandwidth and Colocation Provided
rat.
>
How about this link?
http://blogs.digium.com/2012/11/05/how-to-install-asterisk-11-on-centos-6/
--Satish Barot
Ahmedabad,India
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a l
he dialplan with
>> 'Set(CHANNEL(musicclass)=' or a combination of StartMusicOnHold() and
>> StopMusicOnHold().
>>
>> Can anybody point me in the right direction?
>>
>> Create a script to check for channels on Musiconhold and Originate calls
> throu
On Fri, Jan 11, 2013 at 10:29 AM, Satish Barot wrote:
> On Thu, Jan 10, 2013 at 7:53 PM, RSCL Mumbai wrote:
>
>> Hello,
>>
>> Can asteriskCDR logs tell me if a call was disconnected by the caller
>> or the Agent ?
>>
>> My call flow is as follows:
>
; n,Set(HNGPPARTY=CALLER)
same => n,Queue(QNAME,c,,,60)
same => n,ExecIf($["${QUEUESTATUS}" = "CONTINUE"]?Set(HNGPPARTY=AGENT):)
... ...
exten => h,1,set(CDR(userfield)=${HNGPPARTY})
Note that if nobody answers
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 wrote:
> Hi Satish,
>
> ** **
>
> Thanks for your response – sorry on the slow reply.
>
> ** **
On Mon, Dec 31, 2012 at 3:28 PM, Satish Barot wrote:
> On Mon, Dec 31, 2012 at 3:12 PM, Vinod Nadiadwala wrote:
>
>> Hi,
>>
>> I am new to asterisk, i want to know that is it possible to use asterisk
>> for build voice recording system.
>>
>> Scenario :
sk.
https://wiki.asterisk.org/wiki/display/AST/Application_Record
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
rguments and not the
extension and priority value respectively. Calling Subroutine from dial
will always start execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using
^ as a delimiter.
--Satish Barot
>
>
> ** **
>
&g
**
>
> Thanks all!
>
> ** **
>
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by htt
(DAHDI/g0/${external_num},30)
You can also use Asterisk 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 wrote:
I put ${CHANNEL(dahdi_span)} to know the span and ${CHANNEL(dahdi_channel)}
for actual channel number in incoming context of PRI.
For outbound I normally use M flag in Dial() to call a macro and check the
above variables in that macro.
--Satish Barot
On Tue, Nov 6, 2012 at 7:02 PM, Amit Patkar
es.
>
> Any help really appreciated!
>
> sean
>
Replace your line with this and see..
same=n,GoSubIf($[${CALLERID(**num)} = 2024324321]?other,${**thisexten}:)
--Satish Barot
--
_
-- Bandwidth and Colocation Provided by ht
UBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
Glad I found you asking a question!
Check a function IAXVAR.
I think Asterisk version matters for it.
--Satish Barot
--
_
0048",
> "waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1") in new
> stack
> waiting: 1 calls in queue: 1 avg hold: 0 logged in: 1 ready: 1
>
>
QUEUEHOLDTIME and some other Queue variables will be set just prior to the
caller being bridged with a queue mem
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
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
ameter.
Store the value somewhere in Database)
(5)When you think your Agents 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&#
Yes of course you can use local channel with AddQueueMember().
--Satish Barot
On Wed, Apr 11, 2012 at 1:22 PM, Olivier wrote:
> 2012/4/11, Satish Barot :
> > I would implement it in a different way.
> > As you seem to be a seasoned player just a hint here.
> > How about add
extension rings or not. But at least you can identify which
extension is being dialed.
See 'Using Local Channels' on
http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html
--Satish Barot
On Wed, Apr 11, 2012 at 2:45 AM, Todd Routhier wrote:
> Thanks again Danny, Perl was the
ayesh
>
>
> On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot
> wrote:
>
>> 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 ex
: {your_meetme_number_here}
Hope this helps.
--Satish Barot
On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar wrote:
> 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 123
nel
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)
--Sa
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 wrote:
> Satish,
>
> As if I know, PRI provider give you PRI number at th
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 wrote:
> India TRAI rules doesn't allow for CLID setting. They are backwards
> minded. If y
3/')
MP3DEST="/var/spool/asterisk/mp3/$MP3"
/usr/bin/lame "${WAV}" "${MP3DEST}" --silent -b 16 -s 9.6 -m m --bitwidth 8
--lowpass 9.6 --resample 8 --lowpass-width 1
--SATISH BAROT
Ahmedabad,India.
On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib wrote:
> Hello G
ame => n,Wait(10)
same => n,SendDTMF(3)
Hope this helps you,
--SATISH BAROT
On Wed, Dec 28, 2011 at 3:02 PM, virendra bhati wrote:
> Hi Satish,
>
> Thank you Satish. I did the same before your e-mail i saw. But i got
> another issue in such case.
> DTMF is passed to tha
length of IVR file in
seconds.
same => n,Wait(10)
same => n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati wrote:
> Hi list,
>
> Is there any way in asterisk by which I make a call from server and then
> dialplan(IVR system) gets DTMF from it. I mean
Use callcounter = yes in sip.conf for 1.8
--SATISH
On Wed, Nov 23, 2011 at 11:10 AM, Raj Mathur (राज माथुर) <
r...@linux-delhi.org> wrote:
> On Wednesday 23 Nov 2011, bilal ghayyad wrote:
> > Asterisk version is 1.8.4.2
> >
> > Just I need to know if the below is a normal behaviour of asterisk
Have something like this with necessary changes as per your requirement.
[default]
exten => _X.,1,ExecIf($[$["${CALLERID(num)}" = "667"] | $["${CALLERID(num)}"
=
"667"]]?Goto(isd,${EXTEN},1):Goto(local,${EXTEN},1))
[local]
; Mumbai Mobile Numbers
exten => _9X,1,AGI(agi://127.0.0.1:4577/c
Check 'retry' in queues.conf
[SATISH]
Mumbai, India.
On Sun, Jul 10, 2011 at 4:34 PM, Florent THOMAS 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 ther any param
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, 2
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 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/
; 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
>
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.
--
_
-- Bandwidth and Colocation Provided
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]
Mumb
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 7:03
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)
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 <
of 'qualify' in my 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
> wrote:
>
>>
>> Provide the
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 dis
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 Thu
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 wrote:
> On 06/08/2011 09:10 AM, Satish Barot wrote:
>
>>
>> Set queue_log = no in logger.conf. By default it is set to 'yes'.
>
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 wrote:
> 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
> /var/log/asterisk/que
ALLERID(num)=${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 wrote:
> Sir,
>
> I have MYsql database in myserver.
>
>
> On Tue, Jun 7, 2
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 wrote:
> Hi,
>
> I have small confusion in my configuration which is I had some DID's like
> 044578900-04457999. I was configured dial plan bel
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),
PRI
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 wrote:
> May I add...I still ha
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 submit
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,
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
> Good afternoon,
>
> I'm trying to write a simple callback context, but i need to hangup an
> incoming ca
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 wrote:
> On 06/02/2011 10:29 AM, Eric Wieling wrote:
>
>>
>>
>> -Original Message-
>>> From: asterisk
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
wrote:
> 1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term
>
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 => **0,cal
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 wrote:
>
> I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt
> find this release from asterisk community.
>
>
>
>
> --
> __
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(${EPOCH},,%Y-%m-%d
%H:%M:%S)})
same => n,DBdeltree(${
>
> I have done like this way hope it works for you.
>
> --
> Regards,
>
> Chandrakant Solanki
>
>
> On Mon, May 30, 2011 at 2:53 PM, Satish Barot
> wrote:
>
>> While playing with DB function in Dialplan, I have added some garbage in
>> AstDB. These are
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
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 AM
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 i
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 1.
94 matches
Mail list logo
