Hi Eric Thanks for the pointer.
I've set 'escape display name' to 'yes' and removed the CLD prefix I had added
("T: ") and this is now working. Perhaps it was not escaping the : or space
char from the CLD prefix I had added.
Cheers
Jesse
On 17/02/2010, at 10:35 AM, Eric Varsanyi wrote:
> 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 sipx-users@list.sipfoundry.org
>>>> 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
>>
>
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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