On Mon, 24 Aug 2015 23:48:50 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
When I dial 416555 I get no ringback which I can sort of live with
since it gets answered pretty quickly but then when I dial 200 it
transfers me correctly to the 416555 extension but there is no
ringback there
D'Arcy J.M. Cain wrote:
My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
snip
[pbx-17842]
exten = s,1,Verbose(0,${CALLERID(all)} Calling PBX 17842)
On Tue, 25 Aug 2015 14:51:56 -0300
Joshua Colp jc...@digium.com wrote:
When I dial 416555 I get no ringback which I can sort of live
with since it gets answered pretty quickly but then when I dial
In fact I added a Ringing and a Wait(3) so there is at least one
ringback now.
200 it
On Mon, 24 Aug 2015 23:48:50 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
exten = 200,1,Verbose(0,${CALLERID(all)} Calling PBX darcy)
same = n,GoTo(LocalSets,416555,1)
I tried changing the above to;
same = n,Dial(SIP/416555)
and
same = n,Dial(SIP/416555,,r)
Same problem.
My last problem was nicely solved through this mailing list so
hopefully this new problem will have the same happy outcome.
My situation is that I have many extensions. Here is a sample:
[client-phone](!)
type=friend
host=dynamic
secret=XX
dtmfmode=auto
disallow=all
allow=ulaw
allow=gsm
I would like to configure Asterisk send back only a Trying or Progress message
to the SIP client and not any early audio for ringback. I've confirmed
Asterisk is sending RTP when the call is ringing by using rtp debug on Asterisk.
Does anyone have any ideas on how to accomplish this?
I've
Chris Abel writes:
Hello everyone!
I've had this problem for a while and cant figure it out. When an outside
caller calls an extension on my asterisk system, they do not hear any
sort of ringing. Inside extensions calling other extensions do hear
ringing. We have 3 other asterisk systems that are
Good morning,
i've noticed many times that there are IVRs that play a ring tone just
before bridging me to an agent. My asterisk does not behave like this
but i've always wanted to.
I'm now playing with 1.6.2.9 and i've read in queue's doc:
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ad...@3a.hu
Sent: Friday, July 23, 2010 3:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ringback tone after MOH, before queue member bridged
Good morning,
i've noticed many times that there are IVRs
Hi,
I'm going abroad shortly and want to be able to dial into asterisk and get it
to call me back so that I can make an outgoing call through my voip provider,
rather than paying crazy international rates.
Can anyone point me in the right direction with regards to the dialplan?
Im using
The problem was because my res_indications.so not been loaded.
I added it in my modules.conf and now everithing works fine.
Thanks a lot
2008/9/5 eng. Anatoli Marinov [EMAIL PROTECTED]:
I do not know but I could not set it up. :) bad luck maybe.
2008/9/4 Steve Totaro [EMAIL PROTECTED]:
Why
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
The problem appears when the call comes from external point to our
internal network. So when the server receives the call the channel is
answered and the remote user hears prompt which invite him to
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for inband tones like after an
Answer())
eng. Anatoli Marinov wrote:
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
The problem appears when the call comes
Is there any special option which I should enable to activate these tones?
My progressinband is yes and I cal Dial app with r option it it right?
2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for
This has nothing to do with the progressinband setting and you should
never use the r option.
eng. Anatoli Marinov wrote:
Is there any special option which I should enable to activate these tones?
My progressinband is yes and I cal Dial app with r option it it right?
2008/9/4 Eric
So as I understand the only thing that I can do is to set up
indications.conf. Ok I will try it tomorrow and will write again with
my results.
Thanks a lot.
2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
This has nothing to do with the progressinband setting and you should
never use the
Why is it an option if it should never be used?.
Thanks,
Steve Totaro
On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
This has nothing to do with the progressinband setting and you should
never use the r option.
eng. Anatoli Marinov wrote:
Is there any
I do not know but I could not set it up. :) bad luck maybe.
2008/9/4 Steve Totaro [EMAIL PROTECTED]:
Why is it an option if it should never be used?.
Thanks,
Steve Totaro
On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
This has nothing to do with the
It just might be that your carrier is not sending ring . You can use
'r' in asterisk dial command in extensions.conf to generate ring from
asterisk .
On 31/05/07, dima [EMAIL PROTECTED] wrote:
Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes,
Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes, when making a call, my asterisk box doesn't detect a ringback
and I just hear silence until the other party picks up the phone. I've
checked the SIP messages and they are ok (I'm getting 183 session
Is anyone aware of a method to signal back to a caller that
the party they have called is on another call (other than the obvious -send
to VM or a busy signal).
What I am thinking of is the alternate ring you get when you
call someone on various cellular networks or PSTN networks where
Peter Svensson wrote:
On Thu, 20 Jan 2005, Steve Clark wrote:
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
snip
Thanks Peter,
Hi list,
I am dialing from one zap channel to a second zap channel. Is there a way to
keep the channel I am dialing to from generating a ringback tone.
Sorry if this is obvious but I have been looking for a day trying to find an
example of how to do this.
Thanks,
Steve
--
They that give up
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
should not generate ringback...
exten = 1,Dial(Zap/1,,r)
should generate
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
should not generate ringback...
exten = 1,Dial(Zap/1,,r)
On January 20, 2005 02:24 pm, Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
should not generate
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
should not generate ringback...
exten = 1,Dial(Zap/1,,r)
On Thu, 20 Jan 2005, Steve Clark wrote:
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten = 1,Dial(Zap/1)
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, December 29, 2004 2:47 PM
Subject: [Asterisk-Users] RINGBACK then HANGUP
I am using the manager API to sucessfully ORIGINATE a call. I am using
PHP. I connect to asterisk and then connect
I am using the manager API to sucessfully ORIGINATE a call. I am using PHP.
I connect to asterisk and then connect an internal SIP phone to an external
phone.
?php
$timeout = 7500;
$login_extension = SIP/6001; // agent extension
$call_telephone = 9707; //
On Tue, 2004-03-23 at 09:12, Jeremy Jones wrote:
I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and
bumped into a little problem...
Dialing in from the pstn to sip phones (x-lite softphone on winders and
a grandstream handytone-286 ata), I hear the sip phone ring a few times,
On Tue, 2004-03-23 at 09:49, Eric Wieling wrote:
I'm having a similar problem with 0.7.2 but ONLY if I dial multiple
destinations at the same time. Here is a copy of my extension section
that does NOT provide ringback no matter what I do. In this example the
caller hears ringing while the
I just upgraded to the latest cvs (3/23/04 @ about 7:00am MST) and
bumped into a little problem...
Dialing in from the pstn to sip phones (x-lite softphone on winders and
a grandstream handytone-286 ata), I hear the sip phone ring a few times,
but hear nothing from the pstn side 'til the
I would just like to follow-up on the ringback
problem I'm getting from *. As I've said in my previous post, I am not
hearing the "real ringback" from the Cisco gateway terminating my call. I
don't want to provide false ringback from * (r option of dial), because it'll
still give me
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