Thanks a lot Dona and jg for your inputs.
I'll try to find some way to do this from Dialplan or AMI and let you guys know
soon. Please share if you have some more ideas.
Regards,
Rajib
Date: Tue, 11 Jun 2013 18:34:46 +0200
From: jg
Subject: Re: [asterisk-users] announcement to be played for at
hi,
I've solved various iax2 problem mentioning calltoken when I put
these lines in the iax configuration:
requirecalltoken=no
calltokenoptional=0.0.0.0/0.0.0.0
bye
Il 11/06/2013 19:25, Mordechay Kaganer
scrisse:
B.H.
B.H.
On Jun 11, 2013 5:15 PM, "Steve Totaro"
wrote:
>
>
>
>
> On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer
wrote:
>>
>> B.H.
>>
>> Hello!
>>
>> We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
Hello Adam,
Thank you very much for your info.
Regards,
Jonson.
On Tue, Jun 11, 2013 at 12:34 AM, wrote:
> Hi,
>
>
> On 06/10/2013 22:26, Jonson Player wrote:
>
>> Some users of main use + instead of 00 for international dial. Is there
>> any solution for this problem?
>>
>
> swap the + sign t
How about
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html
?
jg
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New to Asterisk? Join us for a live introductory we
I need to install cdr_mysql.so module for logging call to mysql. I have
the source file cdr_mysql.c only. Can someone explain the steps needed to
get this module compiled and working in Asterisk 1.8.22.0 on CentOS.
Thanks.
Nick
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Since Dial() might not return, DIALSTATUS cannot be used.
I checked the various AMI events and you'll see a bunch of Newchannel,
Hangup, Bridge, Unlink, and Masquerade events when transferring calls.
You could use this to originate a call with the announcement for B. This
is ugly, but if B's p
On Tue, 11 Jun 2013 18:43:55 +0300
Tzafrir Cohen wrote:
> On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
> > This is the second issue I found while trying to install Asterisk
> > on a NetBSD box. I can't load the rtp module because
> > HAVE_OPENSSL_SRTP seems to be set. Is the
So, B transfers the call and after bridging to C, B should get an
announcement.
This is just an idea:
See whether you can dispatch the termination of the call leg B-C by
evaluating the DIALSTATUS variable. I am not sure whether you can see
this inside the dialplan, but you should get the event
On Tue, 11 Jun 2013 18:42:07 +0300
Tzafrir Cohen wrote:
> On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
> > I am trying to build Asterisk on a NetBSD system but I am running
> > into two problems. The first only happens on an installation built
> > from NetBSD HEAD. The confi
Jonas Kellens wrote:
>
> Even if there *can* be more than 1 digit, in case there is only 1 digit it
> should go faster.
Jonas,
Use the TIMEOUT function to set the maximum amount of time permitted between
digits when the user is typing in DTMF. As you've discovered, the default is 5
seconds.
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
> This is the second issue I found while trying to install Asterisk on a
> NetBSD box. I can't load the rtp module because HAVE_OPENSSL_SRTP
> seems to be set. Is there some way to simply force this variab;e to be
> unset from a co
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
> I am trying to build Asterisk on a NetBSD system but I am running into
> two problems. The first only happens on an installation built from
> NetBSD HEAD. The config variable HAVE_NEWLOCALE is erroneously set
> during configure b
On 06/11/2013 04:46 PM, Eric Wieling wrote:
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
I understand that I
No. When you dial "1" the PBX does not know if it needs to match _X or _X.
-Original Message-
From: Jonas Kellens [mailto:jonas.kell...@telenet.be]
Sent: Tuesday, June 11, 2013 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Eric Wieling
Subject: Re: [asteris
On 06/11/2013 04:44 PM, Jonas Kellens wrote:
[snip]
Ok thanks.
Any idea how I can resolve this ?
Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.
Would it help if they pressed for example "1" followed by the "#" key?
If not then, as Eric mentioned,
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
-Original Message-
From: asterisk-users-boun...@lists.digi
On 06/11/2013 04:39 PM, Richard Mudgett wrote:
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher an
Yes, using cdr_adaptive_odbc.conf.
As it is a new table, just changed the name from calldate to start and now
it is inserting the field ok.
Thank you very much for your help.
Best.
2013/6/11 Kevin Larsen
> Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no
> Asterisk ana
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens wrote:
> On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
>
> Jonas Kellens wrote:
>
> I notice that it takes 4 to 6 seconds between someone pressing a cipher and
> Asterisk continuing inside the dialplan. How come ???
>
> ...
>
> Why doesn't Asterisk
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF
On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer wrote:
> B.H.
>
> Hello!
>
> We have several Asterik boxes that are connected to PSTN using PRI cards
> and they are interconnected using IAX2 trunks so that incoming calls are
> delivered from PSTN to the servers they belong to.
>
> In past we we
Jonas Kellens wrote:
>
> I notice that it takes 4 to 6 seconds between someone pressing a cipher and
> Asterisk continuing inside the dialplan. How come ???
>
> ...
>
> Why doesn't Asterisk continue immediately inside the dialplan after having
> received the DTMF-input ?
Jonas,
Please provide
Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no
Asterisk analog to calldate. You would need an alias set up. Mine looks
like:
alias start => calldate
so that the start of my call is what gets logged to the database as the
calldate.
Kevin Larsen
From: Jairo
To:
jg
Sent: Tuesday, June 11, 2013 5:28 AM
Playing an announcement like "Your call has been..." to A after C has
accepted the call is probably not a good idea, because C has to wait until
the the announcement has finished. In environments where callers are
announced to C, C would typically not want t
Hello,
I notice that it takes 4 to 6 seconds between someone pressing a cipher
and Asterisk continuing inside the dialplan. How come ???
Taken from verbose logfile :
(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on
SIP/SipAgenT01-1eb0
[Jun 11 15:29:25] DTMF
Hello,
Still about CDR and MySQL table, should the calldate field be inserted by
Asterisk?
This is the table structure we are using, based on Asterisk wiki:
mysql> describe cdr;
+-+---+--+-+-++
| Field | Type
B.H.
On Tue, Jun 11, 2013 at 3:45 PM, Doug Lytle wrote:
> >> WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
> from address X.X.X.X
>
> I don't know if this will help, but I have:
>
> requirecalltoken=no
>
> In my iax.conf
>
> Doug
>
Thanks, Doug. I too have it there and t
>> WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from
>> address X.X.X.X
I don't know if this will help, but I have:
requirecalltoken=no
In my iax.conf
Doug
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B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.
In past we were using asterisk 1.4 on the server that is receiving IAX
connections and
While B is talking to C, A is enjoying MOH. You could install a
musicclass that starts with "Your are being...".
Playing an announcement like "Your call has been..." to A after C has
accepted the call is probably not a good idea, because C has to wait
until the the announcement has finished. I
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