On Tue, Feb 16, 2010 at 7:05 PM, Eric Varsanyi <[email protected]> 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 [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 > > > > _______________________________________________ > 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/ > This is why I only use Patton gateways. 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