Re: [asterisk-users] Still does not work

[Joshua Colp]

On Fri, Dec 9, 2016, at 02:38 PM, Saint Michael wrote:
> ​Connected to Asterisk 13.13.1 currently running on siptrunks (pid = 336)
> [Dec  9 18:37:34] ERROR[29914]: res_pjsip_sdp_rtp.c:184 create_rtp:
> Unable
> to create RTP instance using RTP engine 'asterisk'​

This was a security release, not a regression release.

-- 
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] Still does not work

[Saint Michael]

​Connected to Asterisk 13.13.1 currently running on siptrunks (pid = 336)
[Dec  9 18:37:34] ERROR[29914]: res_pjsip_sdp_rtp.c:184 create_rtp: Unable
to create RTP instance using RTP engine 'asterisk'​
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Re: [asterisk-users] IMPORT from bridged Local channels not importing.

[Richard Mudgett]

On Tue, Nov 15, 2016 at 8:21 AM, Ethy H. Brito 
wrote:

>
> Hi All
>
> I have some users that can access outside world telephone number.
> They have external numbers to be reached as well.
>
> Due to internal policy restrictions, they are not allowed to dial
> each other internal numbers. I Can't change that.
>
> When an internal user dials the external number for another internal user,
> I Dial(Local/...) the second user.
>
> So I end up with two channels:
> SIP/origin to Local/dest_ext_num;1
> and
> Local/dest_ext_num;2 to SIP/destination
>
> When the call is hung up (h extension), I need to grab the stats of both
> legs (SIP/origin and SIP/destination) of the call, so I use:
>
> ${RTPAUDIOQOS}
>
> to grab the origin leg stats and
>
> Set(MyDESTCH=${CUT(CDR(dstchannel),\;,1)}\;2)
> Set(DESTCH=${IMPORT(${MyDESTCH},BRIDGEPEER)})
> Set(STATS=${IMPORT(${DESTCH},CHANNEL(rtpqos,audio,all))})
>
> to grab the stats for destination leg.
>
> MyDESTCH is correctly set to "Local/dest_ext_num;2"
> DESTCH receives "SIP/destination"
> But STATS is ""
>
> What am I missing here?
>

You cannot get information from a channel that no longer exists.
SIP/destination almost certainly no longer exists when the h exten
runs on SIP/origin.


> Is there a smarter way for grabbing these?
>

I would suggest using hangup handlers [1].  Add a hangup handler to
SIP/origin and SIP/destination.  The handler will run on the respective
channel when it hangs up.  These two channels will hang up at different
times so you can only get the current channel's information.  If you have
to process both channel's information then you are going to have to do that
after both channels have hung up.


>
> Another questions: when the call is hung, in which context is "h
> extension" run?

Always originator? Always destination? Depends on what?
> What about in this scenario I describe (four contexts involved)?
>

In your case, the h exten is run by SIP/origin and Local/dest_ext_num;2.
These
channels executed dialplan when the call was originally placed.  The h
exten runs
on the respective channel in the last context the channel executed
dialplan.  This
is usually at the Dial application.

Richard

[1] https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers
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Re: [asterisk-users] Tranfer the called number in 3 way call

[John Kiniston]

You could use the IAXVAR() function to set variables before dialing your
IAX peer on the initial PBX that then get retrieved by the 2nd PBX.
PBXA:
same => n,Set(IAXVAR(CALLERID)=${CALLERID(num)})
same => n,Set(IAXVAR(DNID)=${CALLERID(DNID)})

PBXB:
same => n,NoOP(CallerID is ${IAXVAR(CALLERID)} DNID is ${IAXVAR(DNID)})


On Thu, Dec 8, 2016 at 10:34 PM, sam habash  wrote:

> Hey there,
>
>
> I have a question i want a dialplan to send the called number of the
> client instead of my callerID when making a 3way call or when transfering
> to an extension from a bridge to another pbx. The problem i add a variable
> and using thw two underscores but i still see the my calledID , i am using
> both asterisk 1.8.29 and other is asterisk 11.4 here is the dialplan
> inherit which i do :
>
>
> exten=> _6XX,n,set(__var=${EXTEN})
>
> exten=> _6XX,n,Dial(IAX2/bridge/${EXTEN},,tTor)
>
>
> I want to recieve the client number that were called from my first pvx
> with 5XX extensions to be shown on my second pbx with 6XX while making a
> transfer to the bridge with 3 way call or blind transfer , I know i am
> missing something here , can you guys help me.
> Sent from my LG Mobile
>
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>
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