date:20160823

Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread Israel Gottlieb

Maybe try progress() instead of answer ()

בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"  כתב:

> Thank you, I just tried your suggestion. Strangely, the announcement is
> played only if I try to dial a SIP peer which is not available (not
> registered to be more precise). If the SIP peer is available, I only get
> the ring tone, and never hear the announcement. Here is the dialplan (I had
> to add an Answer() before the Dial, otherwise the announcement is never
> played, even in the first case) :
>
> exten = 007,1,Answer()
> same  = n,Dial(SIP/foo/s@playme,40)
>
> [playme]
> exten = s,1,Ringing()
> same  = n,Wait(10)
> same  = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer)
> When it is working, I can see the following output in the CLI, which is
> not there otherwise :
> -- SIP/x requested media update control 26, passing it to
> Local/s@playme-05be;1
>
> Otherwise, no error message, Asterisk tells he is playing the announcement
> but I don't hear it.
>
> Best regards
>
> Jean Aunis
>
> Le 23/08/2016 à 16:07, David Duffett a écrit :
>
> How about:
>
> exten => s,1,Dial(SIP/alice/555@delayed-announce,40)
>
> [delayed-announce]
> exten =>  555,1,Wait(20)
> same => n,Playback(myannouncement,noanswer)
> same => n,NoOP(Whatever else you want to do goes here)
>
> The 'noanswer' option on the Playback means that SIP/alice should continue
> to ring for the remaining 20 of the 40 seconds, as the Playback will not
> answer (terminate) the call.
>
> Don't forget AstriCon this year - www.astricon.net
>
> On 23 August 2016 at 12:52, Israel Gottlieb  wrote:
>
>> You could m and make a moh file that has ringing the first 30 sec and
>> then the anouncment
>>
>> בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"  כתב:
>>
>> Thank you for the idea. The problem with RetryDial, is that it will
>>> cancel the first call, play the announce and then dial the SIP peer once
>>> again, so the telephone will display a missed call. I would prefer to do
>>> everything in a single call.
>>>
>>> Le 22/08/2016 à 17:57, John Kiniston a écrit :
>>>
>>> You could try using RetryDial() instead of Dial, It supports playing an
>>> announcement.
>>>
>>>
>>> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis 
>>> wrote:
>>>
 Sorry, I forgot to write that the SIP peer must keep ringing while the
 announcement is being played.

 Le 22/08/2016 à 17:42, John Kiniston a écrit :

 This seems like the obvious answer but maybe I'm misunderstanding the
 question.

 exten => s,1,Dial(SIP/alice,20)
  same =>   n,Playback(myannouncement)
  same =>   n,NoOP(Whatever else you want to do goes here)

 On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis 
 wrote:

> Hello,
>
> I am searching a way to dial a SIP peer, and if it does not answer
> within 20 seconds, play an announcement to the caller. This means that the
> caller would hear a ring tone for 20 seconds, and only then hear the
> announcement if the callee did not answer.
>
> I know it is possible to do this with ARI, but in this particular case
> I do not want to use ARI. I would like to do this purely with dialplan and
> AGI scripts, but I cannot find a way. I have read about the "m" option of
> Dial application, but it starts the announcement immediately, whereas I
> would like to start it after 20 seconds of timeout.
>
> Does anybody have an idea ?
>
> Best regards,
>
> Jean Aunis
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



 --
 A human being should be able to change a diaper, plan an invasion,
 butcher a hog, conn a ship, design a building, write a sonnet, balance
 accounts, build a wall, set a bone, comfort the dying, take orders, give
 orders, cooperate, act alone, solve equations, analyze a new problem, pitch
 manure, program a computer, cook a tasty meal, fight efficiently, die
 gallantly. Specialization is for insects.
 ---Heinlein




 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
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>>>
>>>
>>>
>>> --
>>> A

Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread John Kiniston

Damn, I was going to suggest trying a Queue with a single member using the
'r' option to play ringing instead of MOH and using an announcement but the
queue will stop ringing your agent while it plays the announcement.

It'd go right back to ringing after the announcement however.

On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis  wrote:

> Sorry, I forgot to write that the SIP peer must keep ringing while the
> announcement is being played.
>
> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>
> This seems like the obvious answer but maybe I'm misunderstanding the
> question.
>
> exten => s,1,Dial(SIP/alice,20)
>  same =>   n,Playback(myannouncement)
>  same =>   n,NoOP(Whatever else you want to do goes here)
>
> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis  wrote:
>
>> Hello,
>>
>> I am searching a way to dial a SIP peer, and if it does not answer within
>> 20 seconds, play an announcement to the caller. This means that the caller
>> would hear a ring tone for 20 seconds, and only then hear the announcement
>> if the callee did not answer.
>>
>> I know it is possible to do this with ARI, but in this particular case I
>> do not want to use ARI. I would like to do this purely with dialplan and
>> AGI scripts, but I cannot find a way. I have read about the "m" option of
>> Dial application, but it starts the announcement immediately, whereas I
>> would like to start it after 20 seconds of timeout.
>>
>> Does anybody have an idea ?
>>
>> Best regards,
>>
>> Jean Aunis
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
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Re: [asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

2016-08-23 Thread George Joseph

On Tue, Aug 23, 2016 at 11:19 AM, Jonathan H  wrote:

> Here's a weirdness - I got a call from someone who couldn't get to my info
> line earlier, I tried it and it was busy tone.
>
> Being on a layby beside a road on a mobile on a long journey, my only real
> option was a remote server reboot so I couldn't diagnose further.
>
> That fixed it, but here's the weirdness:
>
> [Aug 23 14:16:59] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
> 'INVITE' from '"thecallerID" ' failed for
> '46.31.X.X:5060' (callid: VF0046af0fc3f12e7dddb70d62974exx@voipfone) - No
> matching endpoint found
> [Aug 23 14:17:25] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
> 'INVITE' from '"thecallerID" ' failed for
> '46.31.X.X:5060' (callid: VFdfc3ae4343390139cb285533bc97xx@voipfone) - No
> matching endpoint found
> [Aug 23 14:18:01] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
> 'INVITE' from '"thecallerID" ' failed for
> '46.31.X.X:5060' (callid: VF9f956b236cef2b7609037335db7xx@voipfone) - No
> matching endpoint found
> [Aug 23 14:28:54] NOTICE[4014] res_pjsip/pjsip_distributor.c: Request
> 'INVITE' from '"thecallerID" ' failed for
> '46.31.X.X:5060' (callid: VFb797ae4a6e90a05a480de806afxxe@voipfone) - No
> matching endpoint found
>

Assuming you have an "identify" object defined for the ITSP, it sounds like
it didn't match for some reason.  You might want to double check the
"match" parameter.  Maybe there's a typo or something and the ITSP just
happened to use one of the new addresses for that particular call.


>
> All I can tell you is that this server has been up and running for 3
> months without problem (occasional reboot).
>
> The host's monitors showed no strange spikes, it didn't run out of memory,
> and I can find nothing in syslog related to Asterisk between a known "good"
> time and the time of the failed calls.
>
> And of course, the Asterisk process didn't crash, otherwise it wouldn't be
> writing to logs.
>
> The only thing that might be relevant is that yesterday,  the ITSP
> increased its IP address ranges, but I already had those ranges listed in
> iptables and pjsip.conf.
>
> And besides, if the traffic hadn't been blocked, then, again, it wouldn't
> have made it through  to Asterisk to get logged. Am I correct?
>
> So, it's like it somehow got amnesia. The only other strange thing I am
> doing is running this script at just after midnight each night, although
> it's been running just fine for a couple of weeks now.
>
> (It's a workaround for a moh fiddle I'm running - see a previous post but
> I can't see it as being relevant).
>
> pkill mpg123 >> /root/tidy.log
> /usr/sbin/asterisk -x "moh reload" >> /root/tidy.log
> /usr/sbin/asterisk -x "database query \"DELETE FROM \`astdb\` WHERE
> \`key\` LIKE \'%Last%\';\"" >> /root/tidy.log
>
> So... it just seems like Asterisk got pjsip endpoint amnesia - is that
> possible? Anyone come across this before?
>
> Thanks.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] Audio cut-outs

2016-08-23 Thread eli vaughan

I had this recently... and i bet if you use wireshark/tcpdump youll see a
dns lookup for the server's own hostname right before the cutout, and audio
again after response is received. quick fix is to add the hosts name and ip
to /etc/hosts

https://issues.asterisk.org/jira/browse/ASTERISK-26280

On Tue, Aug 23, 2016 at 12:20 PM, Brent Davidson <
br...@texascountrytitle.com> wrote:

> I'm having an issue with some Snom 300s on a server running Asterisk
> version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1.  There is *NO NAT*
> involved.  Phones and server are plugged into the same network switch, all
> on the same IP range.  The server is running a Wildcard AEX410 analog card
> with 2 FXO modules receiving incoming analog lines.
>
> Occasionally, in the middle of a call, the audio will drop out for between
> 15 and 20 seconds before suddenly coming back.  I've tried running u-Law as
> the codec and licensed g.729 version 13.0_3.1.7 with exactly the same
> results.  I have tried turning on every logging option I can think of to
> troubleshoot this but have not been able to find a solution.  I'm
> troubleshooting by remote, so haven't been able to run a wireshark capture
> yet.
>
> pings to the phones from the Asterisk server show no packet loss during
> the cut-outs.
>
> Any ideas?
>
> Thanks,
> *Brent Davidson*
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> To UNSUBSCRIBE or update options visit:
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[asterisk-users] Audio cut-outs

2016-08-23 Thread Brent Davidson

I'm having an issue with some Snom 300s on a server running Asterisk 
version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1.  There is _*NO 
NAT*_ involved.  Phones and server are plugged into the same network 
switch, all on the same IP range.  The server is running a Wildcard 
AEX410 analog card with 2 FXO modules receiving incoming analog lines.


Occasionally, in the middle of a call, the audio will drop out for 
between 15 and 20 seconds before suddenly coming back.  I've tried 
running u-Law as the codec and licensed g.729 version 13.0_3.1.7 with 
exactly the same results.  I have tried turning on every logging option 
I can think of to troubleshoot this but have not been able to find a 
solution.  I'm troubleshooting by remote, so haven't been able to run a 
wireshark capture yet.


pings to the phones from the Asterisk server show no packet loss during 
the cut-outs.


Any ideas?

Thanks,
*Brent Davidson*


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[asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

2016-08-23 Thread Jonathan H

Here's a weirdness - I got a call from someone who couldn't get to my info
line earlier, I tried it and it was busy tone.

Being on a layby beside a road on a mobile on a long journey, my only real
option was a remote server reboot so I couldn't diagnose further.

That fixed it, but here's the weirdness:

[Aug 23 14:16:59] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"thecallerID" ' failed for
'46.31.X.X:5060' (callid: VF0046af0fc3f12e7dddb70d62974exx@voipfone) - No
matching endpoint found
[Aug 23 14:17:25] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"thecallerID" ' failed for
'46.31.X.X:5060' (callid: VFdfc3ae4343390139cb285533bc97xx@voipfone) - No
matching endpoint found
[Aug 23 14:18:01] NOTICE[5923] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"thecallerID" ' failed for
'46.31.X.X:5060' (callid: VF9f956b236cef2b7609037335db7xx@voipfone) - No
matching endpoint found
[Aug 23 14:28:54] NOTICE[4014] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"thecallerID" ' failed for
'46.31.X.X:5060' (callid: VFb797ae4a6e90a05a480de806afxxe@voipfone) - No
matching endpoint found

All I can tell you is that this server has been up and running for 3 months
without problem (occasional reboot).

The host's monitors showed no strange spikes, it didn't run out of memory,
and I can find nothing in syslog related to Asterisk between a known "good"
time and the time of the failed calls.

And of course, the Asterisk process didn't crash, otherwise it wouldn't be
writing to logs.

The only thing that might be relevant is that yesterday,  the ITSP
increased its IP address ranges, but I already had those ranges listed in
iptables and pjsip.conf.

And besides, if the traffic hadn't been blocked, then, again, it wouldn't
have made it through  to Asterisk to get logged. Am I correct?

So, it's like it somehow got amnesia. The only other strange thing I am
doing is running this script at just after midnight each night, although
it's been running just fine for a couple of weeks now.

(It's a workaround for a moh fiddle I'm running - see a previous post but I
can't see it as being relevant).

pkill mpg123 >> /root/tidy.log
/usr/sbin/asterisk -x "moh reload" >> /root/tidy.log
/usr/sbin/asterisk -x "database query \"DELETE FROM \`astdb\` WHERE \`key\`
LIKE \'%Last%\';\"" >> /root/tidy.log

So... it just seems like Asterisk got pjsip endpoint amnesia - is that
possible? Anyone come across this before?

Thanks.
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Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread Jean Aunis

Thank you, I just tried your suggestion. Strangely, the announcement is 
played only if I try to dial a SIP peer which is not available (not 
registered to be more precise). If the SIP peer is available, I only get 
the ring tone, and never hear the announcement. Here is the dialplan (I 
had to add an Answer() before the Dial, otherwise the announcement is 
never played, even in the first case) :

exten = 007,1,Answer()
same  = n,Dial(SIP/foo/s@playme,40)

[playme]
exten = s,1,Ringing()
same  = n,Wait(10)
same  = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer)

When it is working, I can see the following output in the CLI, which is 
not there otherwise :
-- SIP/x requested media update control 26, passing it to 
Local/s@playme-05be;1

Otherwise, no error message, Asterisk tells he is playing the 
announcement but I don't hear it.

Best regards

Jean Aunis

Le 23/08/2016 à 16:07, David Duffett a écrit :

How about:

exten => s,1,Dial(SIP/alice/555@delayed-announce,40)

[delayed-announce]
exten =>  555,1,Wait(20)
same => n,Playback(myannouncement,noanswer)
same => n,NoOP(Whatever else you want to do goes here)

The 'noanswer' option on the Playback means that SIP/alice should 
continue to ring for the remaining 20 of the 40 seconds, as the 
Playback will not answer (terminate) the call.

Don't forget AstriCon this year - www.astricon.net 

On 23 August 2016 at 12:52, Israel Gottlieb > wrote:

You could m and make a moh file that has ringing the first 30 sec
and then the anouncment

בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" > כתב:

Thank you for the idea. The problem with RetryDial, is that it
will cancel the first call, play the announce and then dial
the SIP peer once again, so the telephone will display a
missed call. I would prefer to do everything in a single call.

Le 22/08/2016 à 17:57, John Kiniston a écrit :

You could try using RetryDial() instead of Dial, It supports
playing an announcement.

On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis
> wrote:

Sorry, I forgot to write that the SIP peer must keep
ringing while the announcement is being played.

Le 22/08/2016 à 17:42, John Kiniston a écrit :

This seems like the obvious answer but maybe I'm
misunderstanding the question.

exten => s,1,Dial(SIP/alice,20)
 same => n,Playback(myannouncement)
 same =>   n,NoOP(Whatever else you want to do goes here)

On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis
>
wrote:

Hello,

I am searching a way to dial a SIP peer, and if it
does not answer within 20 seconds, play an
announcement to the caller. This means that the
caller would hear a ring tone for 20 seconds, and
only then hear the announcement if the callee did
not answer.

I know it is possible to do this with ARI, but in
this particular case I do not want to use ARI. I
would like to do this purely with dialplan and AGI
scripts, but I cannot find a way. I have read about
the "m" option of Dial application, but it starts
the announcement immediately, whereas I would like
to start it after 20 seconds of timeout.

Does anybody have an idea ?

Best regards,

Jean Aunis

-- 
_

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-- 
A human being should be able to change a diaper, plan an

invasion, butcher a hog, conn a ship, design a building,
write a sonnet, balance accounts, build a wall, set a
bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new
problem, pitch manure, program a computer, cook a tasty
meal, fight efficiently, die gallantly. Specialization
is for insects.
---Heinlein

--

Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread David Duffett

How about:

exten => s,1,Dial(SIP/alice/555@delayed-announce,40)

[delayed-announce]
exten =>  555,1,Wait(20)
same => n,Playback(myannouncement,noanswer)
same => n,NoOP(Whatever else you want to do goes here)

The 'noanswer' option on the Playback means that SIP/alice should continue
to ring for the remaining 20 of the 40 seconds, as the Playback will not
answer (terminate) the call.

Don't forget AstriCon this year - www.astricon.net

On 23 August 2016 at 12:52, Israel Gottlieb  wrote:

> You could m and make a moh file that has ringing the first 30 sec and then
> the anouncment
>
> בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"  כתב:
>
> Thank you for the idea. The problem with RetryDial, is that it will cancel
>> the first call, play the announce and then dial the SIP peer once again, so
>> the telephone will display a missed call. I would prefer to do everything
>> in a single call.
>>
>> Le 22/08/2016 à 17:57, John Kiniston a écrit :
>>
>> You could try using RetryDial() instead of Dial, It supports playing an
>> announcement.
>>
>>
>> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis 
>> wrote:
>>
>>> Sorry, I forgot to write that the SIP peer must keep ringing while the
>>> announcement is being played.
>>>
>>> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>>>
>>> This seems like the obvious answer but maybe I'm misunderstanding the
>>> question.
>>>
>>> exten => s,1,Dial(SIP/alice,20)
>>>  same =>   n,Playback(myannouncement)
>>>  same =>   n,NoOP(Whatever else you want to do goes here)
>>>
>>> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis 
>>> wrote:
>>>
 Hello,

 I am searching a way to dial a SIP peer, and if it does not answer
 within 20 seconds, play an announcement to the caller. This means that the
 caller would hear a ring tone for 20 seconds, and only then hear the
 announcement if the callee did not answer.

 I know it is possible to do this with ARI, but in this particular case
 I do not want to use ARI. I would like to do this purely with dialplan and
 AGI scripts, but I cannot find a way. I have read about the "m" option of
 Dial application, but it starts the announcement immediately, whereas I
 would like to start it after 20 seconds of timeout.

 Does anybody have an idea ?

 Best regards,

 Jean Aunis


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>>>
>>>
>>>
>>> --
>>> A human being should be able to change a diaper, plan an invasion,
>>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>>> manure, program a computer, cook a tasty meal, fight efficiently, die
>>> gallantly. Specialization is for insects.
>>> ---Heinlein
>>>
>>>
>>>
>>>
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>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>
>>
>>
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
> New to Asterisk? Start here:
>

Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread Israel Gottlieb

You could m and make a moh file that has ringing the first 30 sec and then
the anouncment

בתאריך 22 באוג׳ 2016 7:19 PM,‏ "Jean Aunis"  כתב:

> Thank you for the idea. The problem with RetryDial, is that it will cancel
> the first call, play the announce and then dial the SIP peer once again, so
> the telephone will display a missed call. I would prefer to do everything
> in a single call.
>
> Le 22/08/2016 à 17:57, John Kiniston a écrit :
>
> You could try using RetryDial() instead of Dial, It supports playing an
> announcement.
>
>
> On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis  wrote:
>
>> Sorry, I forgot to write that the SIP peer must keep ringing while the
>> announcement is being played.
>>
>> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>>
>> This seems like the obvious answer but maybe I'm misunderstanding the
>> question.
>>
>> exten => s,1,Dial(SIP/alice,20)
>>  same =>   n,Playback(myannouncement)
>>  same =>   n,NoOP(Whatever else you want to do goes here)
>>
>> On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis 
>> wrote:
>>
>>> Hello,
>>>
>>> I am searching a way to dial a SIP peer, and if it does not answer
>>> within 20 seconds, play an announcement to the caller. This means that the
>>> caller would hear a ring tone for 20 seconds, and only then hear the
>>> announcement if the callee did not answer.
>>>
>>> I know it is possible to do this with ARI, but in this particular case I
>>> do not want to use ARI. I would like to do this purely with dialplan and
>>> AGI scripts, but I cannot find a way. I have read about the "m" option of
>>> Dial application, but it starts the announcement immediately, whereas I
>>> would like to start it after 20 seconds of timeout.
>>>
>>> Does anybody have an idea ?
>>>
>>> Best regards,
>>>
>>> Jean Aunis
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> A human being should be able to change a diaper, plan an invasion,
>> butcher a hog, conn a ship, design a building, write a sonnet, balance
>> accounts, build a wall, set a bone, comfort the dying, take orders, give
>> orders, cooperate, act alone, solve equations, analyze a new problem, pitch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk Realtime RTUPDATE issue

2016-08-23 Thread Joshua Colp


Ahmed Munir wrote:

Hi,

I'm currently using Asterisk 11.7.0.The issue currently I'm facing in
Asterisk realtime sip_buddies table i.e. if I try to unregister the
extension, ipaddr, port, regseconds, fullcontact, useragent and lastms
remain still populated with data unless do the sip reload. This issue
also obser

In sip.conf the parameter I've enabled/uncommented  for realtime are
only 'rtcachefriends=yes' and rest of the realtime parameters are
commented (set as default).

Please advise, what I'm may missed out.


How exactly are you trying to unregister? A REGISTER with 0 Expires? 
What does the console show?


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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Re: [asterisk-users] Dial and start music on hold after timeout

Re: [asterisk-users] Dial and start music on hold after timeout

Re: [asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

Re: [asterisk-users] Audio cut-outs

[asterisk-users] Audio cut-outs

[asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

Re: [asterisk-users] Dial and start music on hold after timeout

Re: [asterisk-users] Dial and start music on hold after timeout

Re: [asterisk-users] Dial and start music on hold after timeout

Re: [asterisk-users] Asterisk Realtime RTUPDATE issue

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