The firmware for the SP3102 has a bug (longstanding) that makes it put out 
corrupted (non sip compliant) headers. It used to be this crashed Freeswitch 
with a SEGV but that's been fixed post 4.0.4.

This *might* be your issue, you can test by putting in a manual caller id in 
the SPA configuration (set PSTN CID For VoIP CID to 'no' in PSTN to VOIP 
settings, and something like 'PSTN FXO' in Display Name under Subscriber 
information) and telling it to properly quote it (on the SIP tab, SIP 
Parameters page set 'Escape Display Name' to 'yes'). If this makes the static 
CID show up in voicemail then you are being bitten by the firmware bug (the 
firmware bug is that the display name is NOT quoted even with 'Escape display 
name' set to yes when it comes off the PSTN, if there are spaces or other 
special characters the header is no longer sip compliant).

-Eric

On Feb 16, 2010, at 3:25 PM, Jesse Reynolds wrote:

> Thanks Eric and Tony
> 
> On the SPA-3000 I have enabled "Detect Disconnect Tone" and set the 
> Disconnect Tone to the one given at the following web page for Australia / 
> Telstra PSTN lines:
> 
>    http://www.voip-info.org/wiki/view/Sipura+3000
> 
> Namely:
> 
>    4...@-30,4...@-30;1(.375/.375/1+2)
> 
> This has fixed it. 
> 
> Next problem, voicemail emails don't show the caller ID, though caller ID is 
> correctly shown on our Snom voip phones. I'll do some tracing to see what's 
> happening with the From. Presumably this is supposed to work OK?
> 
> Cheers
> Jesse
> 
> On 17/02/2010, at 6:26 AM, Eric Varsanyi wrote:
> 
>> I had this problem too with an SPA 3102. I could never get the linksys 
>> firmware to properly detect CPC via the normal telco means (battery reversal 
>> or loop drop). On the line I was using I could watch with a storage scope 
>> and see the loop break for about 500ms but the SPA would just ignore it no 
>> matter how I tried to configure it. A plain old panasonic answering machine 
>> and a KXTA1232 key system had no problem detecting CPC on the same line.
>> 
>> I recorded a bit of my telco's reorder tone and ran it through Audacity to 
>> find the duration and frequencies, then created a 'tone script' (mine ended 
>> up as "4...@-30,6...@-30;4(.50/.50/1+2)" ) for the SPA so it could detect 
>> call completion based on the busy signal. You also have to set 'Detect 
>> Disconnect Tone' if you use this method.
>> 
>> This works fine but you hear a little reorder tone at the end of each 
>> voicemail, not a big deal.
>> 
>> If your telco doesn't provide a reorder tone when a caller hangs up there's 
>> also an option (at least on the 3102) to detect 'PSTN long silence', that 
>> might work for you too.
>> 
>> IMO the Linksys/Cisco firmware for this product line is abandoned and buggy.
>> 
>> -Eric Varsanyi
>> 
>> On Feb 16, 2010, at 2:09 AM, Jesse Reynolds wrote:
>> 
>>> Hello
>>> 
>>> I've set up a small SIPX setup at home, to have a play with it really and 
>>> to 'unify' incoming calls via disparate means (voip and pstn) so they can 
>>> be answered on the same set of phones. We're also about to switch to sipX 
>>> at work. 
>>> 
>>> So, the problem I'm having is that when a call comes in on the PSTN, and 
>>> rings out and goes to voicemail, there is always a five minute voicemail 
>>> recorded (with most of it silence) resulting in a 6MB email attachment. 
>>> Furthermore, the PSTN line is tied up for this five minutes even though the 
>>> caller has long since hung up. 
>>> 
>>> I'm using a Sipura SPA-3000 as the PSTN gateway. It registers as extension 
>>> 203 and routes calls to 301, which is a call hunt group (rings all phones). 
>>> 
>>> After the caller has hung up, and before the PSTN line gets freed up, the 
>>> Active call list shows no calls active. 
>>> 
>>> Does anyone have any ideas how I can fix this so it hangs up the PSTN line 
>>> when the caller disconnects, and the voicemail stops recording? 
>>> 
>>> Note also that if the PSTN call is answered by one of our voip phones, and 
>>> both parties hang up, then the PSTN line is freed up. Does the Voicemail 
>>> system need to be told to hang up after a certain amount of silence, eg 10 
>>> seconds? 
>>> 
>>> Thanks very much
>>> Jesse
>>> 
>>>  Jesse Reynolds
>>>  Virtual Artists Pty Ltd - http://www.va.com.au/
>>>  Phone: 08 7120 7134 (Adelaide) or 02 9043 2288 (Sydney)   Mobile: 0414 669 
>>> 790
>>> 
>>> _______________________________________________
>>> sipx-users mailing list [email protected]
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users
>>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
>>> sipXecs IP PBX -- http://www.sipfoundry.org/
>> 
> 
>   Jesse Reynolds
>   Virtual Artists Pty Ltd - http://www.va.com.au/
>   Phone: 08 7120 7134 (Adelaide) or 02 9043 2288 (Sydney)   Mobile: 0414 669 
> 790
> 

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