[asterisk-dev] Asterisk REINVITE

[Kirill Marchuk]

Hi

I've seen exactly the same behaviour and even used gdb breakpoints to 
understand why is this happening (the only mention-worthy difference in 
SIP/SDP between INVITE and re-INVITE was the ;tag added to To: header)

Unfortunately, I did not save the results, but if I remember correctly, 
that happened simply because a channel was added to a bridge, and bridge 
was calling "update_connectedline" function on every of the channels 
involved (including the newly added channel itself)

That was the most basic case we did with ARI, so we were a little 
surprised of course, but somehow we've decided that this is "how ARI 
works" so we stopped further research on this.

Kirill

26.04.2016 21:57, Nitesh Bansal пишет:
> Hi,
>
> c-line in SDP remains the same, only SDP version in the o-line changes.
>
> Thanks,
> Nitesh
>
> On Tue, Apr 26, 2016 at 4:45 PM, Joshua Colp <jcolp at digium.com 
> <mailto:jcolp at digium.com>> wrote:
>
>     Nitesh Bansal wrote:
>
>         Hello,
>
>         I'm building an ARI based conference with Asterisk 13.
>
>         Scenario:
>         Peer A dials into Asterisk, mixing bridge is created and
>         channel 1 put
>         into the bridge.
>         Asterisk is also told to initiate call to a recording server, so
>         recording server is
>         also added into the bridge.
>         I have noticed that after the initial INVITE completes with the
>         Recording Server,
>         Asterisk is doing a REINVITE towards Recording server, this
>         REINVITE has the
>         same media  IP, media port though SDP version number increases.
>
>         I'm really curious why is Asterisk sending this REINVITE on
>         the outbound
>         leg to
>         the Recording server.
>         Any logical rational for doing that?
>
>
>     Is it updating connected line information?
>
>     -- 
>     Joshua Colp
>     Digium, Inc. | Senior Software Developer
>     445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>     Check us out at: www.digium.com <http://www.digium.com> &
>     www.asterisk.org <http://www.asterisk.org>
>
>
>     -- 
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>     asterisk-dev mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20160427/c0694bad/attachment.html>