[Asterisk-Users] Anyone got their Pulver WiSIP phone working with *?

[Mark Phillips]

I have it working!! Caused me to do major changes to * though.

I have had to force all the connected SIP clients to use ULAW rather than
their choice of GSM, ULAW or Speex.

DTMF has to be set to RFC2833 on * and "outband" on WiSIP even though
WiSIP thinks that Inband is RFC2833. I put a call into Pulver about this
and his techy named "Yan" claims this is correct.

I've also found that the Display Name gets mangled. The WiSIP adds a +
symbol where there should be a space so my name becomes Mark+Phillips.
again their techy thinks this is normal.

Thanks to those that gave me clues.

Mark


Mark Phillips said:
> Nope, not nat'd. its on my internal network. I have canreinvite=no set and
> still nothing.
>
> By your response I guess you don't actually own one of these?
>
> Mark
>
>
> Steven Sokol said:
>> Are you natted (behind a NAT screen)?
>>
>> Do you have any other SIP devices connected to the same network segment
>> as
>> your WiFi Access Point?  Are they working correctly?
>>
>> I would try turning off the canreinvite (set = 0) and, if there's a NAT
>> between you and the Asterisk, turn on nat support (nat = yes).
>>
>> Good luck,
>>
>> Steve
>>
>>> -----Original Message-----
>>> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
>>> admin at lists.digium.com] On Behalf Of Mark Phillips
>>> Sent: Tuesday, March 16, 2004 3:04 PM
>>> To: asterisk-users at lists.digium.com
>>> Subject: [Asterisk-Users] Anyone got their Pulver WiSIP phone working
>>> with
>>> *?
>>>
>>> Hi Folks,
>>>
>>> Took delivery of 3 of these today and am having problems. Pulver tech
>>> support is pretty much non existant so I came here.
>>>
>>> Problem is this. Phone configured to talk to * as extn 3405. * shows
>>> that
>>> 3405 has registered and a sip show peers reveals its ip address etc.
>>> Make
>>> a call from the WiSIP to any extension and * shows the call getting as
>>> far
>>> as the native bridge. call fails after that point. If I let the VM pick
>>> up
>>> * plays the VM messages and then hangs up. At no time does the WiSIP
>>> think
>>> its connected.
>>>
>>> In the reverse direction, a call to the WiSIP gets dumped into VM
>>> straight
>>> away.
>>>
>>> If I call the phone by its URI it rings but the calling phone never
>>> thinks
>>> the call has been answered by the WiSIP.
>>>
>>> I have 711 as the codec and inband DTMF (which the WiSIP thinks should
>>> be
>>> 2833).
>>>
>>> Any ideas ...
>>>
>>> Thanks
>>>
>>>
>>> --
>>> Mark Phillips, G7LTT/KC2ENI
>>> Randolph, NJ
>>> http://www.g7ltt.com/
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>


-- 
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/