[Spce-user] mr3.7.1 and conferences

[Andrew Pogrebennyk]

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Hi Marc,
indeed, there has been some reorganization going on in mr3.7.1 around
the voice prompts. We have removed the ngcp-asterisk-sounds and
ngcp-sems-prompts and now provide all voice prompts in ngcp-prompts
package. But the conference prompts need to be uploaded now via the
web interface in the Sound Sets page. Also you need to set the PIN for
the conference in the subscriber preferences.

Actually now I'm thinking that we should provide the old conference
app for the backward compatibility with those subscribers that do not
have the sound sets or the PIN set. A simple switch in config.yml
would do it. For now you may change your proxy.cfg.customtt.tt2 if you
want to rollback to the old conference module: search for #!ifdef
ENABLE_CONFERENCE and change the block below to the following:

                if(uri =~ "@conference\.local$")
                {
                        route(ROUTE_CHECK_PBX);
                        xlog("L_NOTICE", "Conference request - [%
logreq -%]\n");
                        if(is_present_hf("P-App-Name"))
                        {
                                remove_hf("P-App-Name");
                        }
                        append_hf("P-App-Name: conference\r\n");
                        t_on_failure("FAILURE_ROUTE_APPSRV");
                        route(ROUTE_TO_APPSRV);
                        [% debug_dump('exit', 'ROUTE_FIND_CALLEE') %]
                        exit;
                }

Regards,
Andrew

On 01/21/2015 12:14 PM, Marc Storck wrote:
> Good afternoon,
> 
> yesterday I upgraded to mr3.7.1.
> 
> I’ve now noticed that the conferences are inaccessible.
> kamailio-proxy logs:
> 
> Failure route for hunt call -
> R=sip:conf=1234567890 at conference.local ID=3996539423 at 1_2_3_4
> 
> The corresponding SIP message is:
> 
> U 2015/01/21 12:09:32.570559 127.0.0.1:5062 -> 127.0.0.1:5060 
> SIP/2.0 480 Unavailable' Record-Route:
> <sip:127.0.0.1:5062;lr=on;ftag=440842921;did=9fb.4351>' 
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=440842921;ngcplb=yes;socket=udp:1.2.3.1:5060>'
>
> 
Record-Route:
<sip:1.2.3.1;r2=on;lr=on;ftag=440842921;ngcplb=yes;socket=udp:1.2.3.1:5060>'
> Via: SIP/2.0/UDP
> 127.0.0.1;branch=z9hG4bKe2c9.4ec972ccb4eeee93b3820d925763f76a.0' 
> Via: SIP/2.0/UDP
> 1.2.3.4:2057;branch=z9hG4bK4f26742bf11eff6c889e87c465bad5ae;rport=2057'
>
> 
From: "calling-party" <sip:calling-party at sld.tld>;tag=440842921'
> To:
> <sip:called-party at sld.tld;user=phone>;tag=4D2384F1-54BF88EC000897E0-0AFC0700'
>
> 
Call-ID: 3489192731 at 1_2_3_4'
> CSeq: 3 INVITE' Content-Length: 0’
> 
> Unfortunately I found no other clues. I hope someone here can point
> me into the right direction.
> 
> Regards,
> 
> Marc
> 
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