[Spce-user] Upper Registration to Asterisk - transfers not working

[Daniel Grotti]

That's weird,
try to debug on the asterisk side. Is there the RTP connection during
MOH between asterisk and SCPE ? Check with tcpdump or asterisk's "rtp
debug" command.

Daniel





On 02/18/2013 05:25 PM, Theo wrote:
> Hi Daniel
>
> We're using it as an SBC only at the moment with upper registration.
> Asterisk does play the holding music, but it just doesn't come through
> on the phones. If the phones are connected directly to asterisk it
> works fine.
>
> On Mon, Feb 18, 2013 at 6:17 PM, Daniel Grotti <dgrotti at sipwise.com
> <mailto:dgrotti at sipwise.com>> wrote:
>
>     Sorry Theo,
>     my mistake, "music_on_hold" variable is supported only on PRO version.
>     In your case, if every calls pass through asterisk, asterisk
>     should take care about MOH.
>
>     Daniel
>
>
>
>
>
>     On 02/18/2013 04:59 PM, Daniel Grotti wrote:
>>     Hi,
>>     try to add your music on hold file through the Sounds Set Section
>>     (look for "music_on_hold").
>>     Then apply the sound set to your domain.
>>     No music on hold are setted by default.
>>
>>
>>     Daniel
>>
>>
>>
>>
>>
>>
>>
>>     On 02/18/2013 04:50 PM, Theo wrote:
>>>     All good now - the mistake was in the kamailio.cfg - didn't add
>>>     refer properly.
>>>
>>>     I notice that the holding music that we normally get when on
>>>     hold doesn't play. Is there a way to make that work?
>>>
>>>     On Mon, Feb 18, 2013 at 11:29 AM, Theo
>>>     <axessofficetheo at gmail.com <mailto:axessofficetheo at gmail.com>>
>>>     wrote:
>>>
>>>         Ooops, 
>>>
>>>         No, not working as yet. I didn't realise that the phone I
>>>         was using had been reprogrammed to use ## as per Asterisk :-(
>>>
>>>         Will play with the whitelist a bit more, but if anybody
>>>         spots the problem, feel free to let me know :-)
>>>
>>>
>>>         On Mon, Feb 18, 2013 at 11:17 AM, Theo
>>>         <axessofficetheo at gmail.com
>>>         <mailto:axessofficetheo at gmail.com>> wrote:
>>>
>>>             Hi
>>>
>>>             Working! This is what I have as my whitelist now - in
>>>             case anybody needs it, (and in case anybody spots
>>>             something wrong)
>>>
>>>             header_list=P-R-Uri,P-D-Uri,P-Preferred-Identity,P-Asserted-Identity,Diversion,Privacy,Allow,Supported,Require,RAck,P-Out-Socket[%
>>>             IF kamailio.proxy.presence.enable == "yes"
>>>             %],Expires,Accept,Refer-To,Subscription-State,Event[% END %]
>>>
>>>             Cheers
>>>
>>>             On Sat, Feb 16, 2013 at 5:57 PM,
>>>             <marcin.kowalczyk at architekci.it
>>>             <mailto:marcin.kowalczyk at architekci.it>> wrote:
>>>
>>>                 Hi,
>>>
>>>                  ## works, as it's transfer executed by asterisk
>>>                 that came from rtp/dtmf analysys. It will work as
>>>                 long there will be media flow via asterisk
>>>                 (canreinvite=yes in older asterisk releases,
>>>                 directmedia in newer).
>>>
>>>                 Sent from my BlackBerry® smartphone
>>>
>>>                 -----Original Message-----
>>>                 From: Theo <axessofficetheo at gmail.com
>>>                 <mailto:axessofficetheo at gmail.com>>
>>>                 Sender: spce-user-bounces at lists.sipwise.com
>>>                 <mailto:spce-user-bounces at lists.sipwise.com>
>>>                 Date: Sat, 16 Feb 2013 15:30:58
>>>                 To: Jon Bonilla<jbonilla at sipwise.com
>>>                 <mailto:jbonilla at sipwise.com>>
>>>                 Cc: <spce-user at lists.sipwise.com
>>>                 <mailto:spce-user at lists.sipwise.com>>
>>>                 Subject: Re: [Spce-user] Upper Registration to
>>>                 Asterisk - transfers not
>>>                         working
>>>
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>>>
>>>
>>>
>>>
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>>
>>
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