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

[Theo]

Hi

Real life took over for a little bit, back at this now. It's not turning
out to be as easy as I had hoped :-(

I have 2 issues currently with this.

1. SNOM 300 - blind transfer works fine but attended transfer doesn't. It
does send the refer-to and using ngrep I don't see a difference between the
two, but then again I am usually wrong. The attended transfer just seems to
put me on hold and not get out of that.

2. I have another phone for testing - a Fanvil. Transferring connected
directly to asterisk works fine, but through SCPE it doesn't - error
message "refer-to missing"

How do I troubleshoot this further?

On Mon, Feb 18, 2013 at 6:40 PM, Daniel Grotti <dgrotti at sipwise.com> wrote:

>  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>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> 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>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>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>
>>>>> Sender: spce-user-bounces at lists.sipwise.com
>>>>> Date: Sat, 16 Feb 2013 15:30:58
>>>>> To: Jon Bonilla<jbonilla at sipwise.com>
>>>>> Cc: <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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