Re: [asterisk-users] retry loop in ansible ?
Hi,
> Am 08.01.2024 um 18:16 schrieb C. Maj :
>
> On 12/6/23 02:08, Axel Rau wrote:
>> I have a simple config with some phones ringing simultaneously.
>> Some of them are softphones (zoiper apps on iPhone w/o push notification).
>> If such an app did bot register in time, it has no chance to pick up the
>> call.
>> If I could configure a retry loop checking for registered candidates,
>> say once a second until one phone takes the call, this would allow me
>> to pick up the call with zoiper app registered late.
>> How could this be done in ansible?
>
> Did you mean asterisk ?
Yes.
>
> If so, then you might look into the While()/EndWhile() applications, combined
> with timeouts to Dial() application, starting with something very basic such
> as the following:
>
> same = n,Set(tries=0)
> same = n,While($[${INC(tries)}<99])
> same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(1234)})
> same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(5678)})
> same = n,Dial(${team},10)
> same = n,Wait(1)
> same = n,EndWhile()
>
> ...at most that would be 11 seconds in between registration of x5678 and the
> next time it gets called when x1234 is not answering.
Thanks a lot for this example.
>
> Other approaches might involve Queue()'s with some ChannelRedirect()'s or
> even Bridge()'s, maybe AGI/ARI, etc.
>
> BTW the Asterisk Forums are a great place to post these kinds of questions in
> the future: https://community.asterisk.org
I will try this in the future,
Regards, Axel
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Re: [asterisk-users] retry loop in ansible ?
On 12/6/23 02:08, Axel Rau wrote:
I have a simple config with some phones ringing simultaneously.
Some of them are softphones (zoiper apps on iPhone w/o push notification).
If such an app did bot register in time, it has no chance to pick up the call.
If I could configure a retry loop checking for registered candidates,
say once a second until one phone takes the call, this would allow me
to pick up the call with zoiper app registered late.
How could this be done in ansible?
Did you mean asterisk ?
If so, then you might look into the While()/EndWhile() applications,
combined with timeouts to Dial() application, starting with something
very basic such as the following:
same = n,Set(tries=0)
same = n,While($[${INC(tries)}<99])
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(1234)})
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(5678)})
same = n,Dial(${team},10)
same = n,Wait(1)
same = n,EndWhile()
...at most that would be 11 seconds in between registration of x5678 and
the next time it gets called when x1234 is not answering.
Other approaches might involve Queue()'s with some ChannelRedirect()'s
or even Bridge()'s, maybe AGI/ARI, etc.
BTW the Asterisk Forums are a great place to post these kinds of
questions in the future: https://community.asterisk.org
Regards,
--
鸞 C. Maj, TechnoCaptain
Penguin PBX Solutions
Denver 720-32-42-72-9
Beyond 1-833-PNGN-PBX
http://PeNGuiNPBX.com
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Re: [asterisk-users] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip
On Mon, Jan 8, 2024 at 12:07 PM marek wrote: > hi, > > we are moving our asterisk from chan_sip to chan_pjsip > > we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from > REFER (asterisk - other pbbx - SIP REFER - asterisk) > > > https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc82092170e > > is it supported in pjsip too? or is there other way? > Looking at the REFER implementation[1] it seems like no. You can submit a feature request here[2]. [1] https://github.com/asterisk/asterisk/blob/20/res/res_pjsip_refer.c [2] https://github.com/asterisk/asterisk-feature-requests -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip
hi, we are moving our asterisk from chan_sip to chan_pjsip we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from REFER (asterisk - other pbbx - SIP REFER - asterisk) https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc82092170e is it supported in pjsip too? or is there other way? thanks Marek -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
