Re: [asterisk-users] Is it possible to set a timeout when querying a calendar ?

[Ludovic Gasc]

2016-06-21 12:18 GMT+02:00 Olivier :

> Hello,
>
> Asterisk offers several ways to query an external calendar.
> I'm wondering what could happen if an external calendar was down or very
> slow to respond.
>
> Looking at wiki.asterisk.org, I didn't find any timeout parameter in
> calendar functions, nor in calendar.conf (but I'm not too sure for the
> later case) ?
>
> What would you suggest to protect your dialplan from a mis-behaving
> external calendar ?
>

Maybe somebody else has a better pattern, but for now, I've implemented a
middleware to fetch ics calendar and store that in folders served by a
Nginx.
Asterisk retrieves ical directly on localhost.

Two advantages:
1. Asterisk don't waste CPU time to retrieve data.
2. If the calendar source is down, it's invisible from Asterisk.

However, the main issue is that I didn't find a way to force Asterisk to
purge its cache and reload a specific calendar.
I play with timeouts to avoid to be too much in delay with the latest
version available of ical.

Have a nice day.


>
> Best regards
>
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Re: [asterisk-users] I am facing an issue with hep packet sending .

[Joshua Colp]

Sasmita Panda wrote:

Hi ,

 I have installed asterisk-13.9 with pjsip and hep enable . My
asterisk is running . I am able  to get calls . My hep.conf looks like
bellow

[general]
enabled = yes
capture_address = 1.1.x.x:9060
;capture_password = 1234
;capture_id = 1234

 My homer capture server is running in the same machine with 9060
port where asterisk is running in 5080 port . BUt asterisk is not
sending any hep packet to homer capture server .

   rather than hep.conf , am I need to do anything else ? Please
help me . I am middle of something .


Do you have the res_hep.so, res_hep_rtcp.so, and res_hep_pjsip.so 
modules loaded? Are you using PJSIP? Are there any warnings/errors at 
startup about it?


--
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] PJSIP/Realtime RLS

[Joshua Colp]

Kevin Miller wrote:

I see that you can configure RLS in pjsip.conf, but does this work with
realtime? The wiki refers to pjsip.conf for configuration, but since
many of the other items can be in the the DB, I was wondering if RLS can
as well.


There's nothing to stop it from working, and the underlying sorcery 
should allow it... but it's not something that has been tested.


--
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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[asterisk-users] CDR(uniqueid) for attended transfer

[Дорофеев Сергей]

Hello list.

When i have an incoming call with attended transfer, im getting two separate 
records in my CDR with 2 different uniqueid's. Do I have any way to  postpone 
my uniqueid from one call to another?
Im using SIP on Asterisk 11.19 right now.

WBR,

Dorofeev Sergey


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Re: [asterisk-users] Authentication header in BYE packets

[Faheem Muhammad]

Strange, A BYE should be replied with 200 OK, 481 (non matching dialogid),
408 request time out or similar responses, but it should never be
challenged. Only INVITE, REGISTER and  PUBLISH requests are challenged with
401/407.
As per rfc3261 it should not challenge the BYE Requests.
*The workaround is to add a SIP Proxy(opensips/kamillio) in between your
Provider and Asterisk server and manipulate the BYE message with challenge.

Regards,
Muhammad Faheem


On Thu, Jun 23, 2016 at 12:19 AM, Owais Ahmad 
wrote:

> Hi all,
>
> My provider proxy expects authentication header on BYE packets as well. Is
> there a way in asterisk to add this header on BYE packets?
>
> When proxy replies with a 401 on BYE, asterisk just retransmits the BYE
> packet.
>
> Regards,
> Owais
>
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