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Unread 12-29-2007, 10:15 PM   #21
Kenthurst35
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This config is currently set up for routing via the PSTN gateway.

I did have those CFwd parameters on Line 1 set to YES originally, but they didn't seem to have any effect since I was using the gateway method to divert to VM.

Happy to reset them to YES if you think it will make a difference.

Cheers
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Unread 12-29-2007, 10:16 PM   #22
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BTW, I don't give out my VoIP DID, so inbound calls are always via PSTN.

Cheers
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Unread 12-29-2007, 10:26 PM   #23
dc2007
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can you make your vm ringing duration lower than your spa3102 PSTN Answer Delay?
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Unread 12-29-2007, 10:37 PM   #24
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The current settings are:

In the Voxalot account, voicemail ringing duration is set to 5 secs. In the SPA3102 the PSTN Answer Delay is set to 10 secs.

With these settings, my understanding of the call flow is this:

PSTN call --> Line 1 --> ring for duration set in PSTN Answer Delay (10) --> gateway takes call & dials voxalot account --> ring for Voxalot voicemail ringing duration (5) --> VM

Cheers
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Unread 01-02-2008, 09:14 AM   #25
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Anyone out there, with some more ideas on what could be the problem here?

I really am at my wits end with this.
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Unread 01-04-2008, 01:16 PM   #26
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Quote:
Originally Posted by Kenthurst35 View Post
Anyone out there, with some more ideas on what could be the problem here?

I really am at my wits end with this.
I have one more thing for you to try. It solved some problems for me but I'm not sure it is related to your problem. Enable STUN (I use stun.voxalot.com:3478), enable NAT mapping and NAT keep alive. Disable any port forwarding you have in place between the ATA and the router.

Might pay to re-boot everything as well.
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Last edited by ozimarco; 01-04-2008 at 01:19 PM.
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Unread 01-05-2008, 03:09 AM   #27
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Thanks Ozimarco

I'll try using the STUN settings as you suggest and see what happens.

Thing is, if I can make SIP calls to other addresses successfully, wouldn't that indicate that I don't have a STUN problem?

Cheers
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Unread 01-05-2008, 03:19 AM   #28
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OK, I tried changing the STUN settings, but no joy.

Here is the syslog trace:

syslog server(port:514) started on Sat Jan 05 14:11:46 2008
TP Parser error: 34
FXO:Start CNDD
Calling:201259@127.0.0.1:5060
[1:0]AUD ALLOC CALL (port=16384)
[1:0]RTP Rx Up
[0:0]AUD ALLOC CALL (port=16386)
[0:0]RTP Rx Up
CC:Ringback
AUD:Play PSTN Tone 9
[1:0]RTP Rx Dn
FXO:CNDD name=, number=
FXO:Stop CNDD
FXO:CNDD Name= Phone=
AUD:Stop PSTN Tone
[5062]STUN trying 0
[16388]STUN trying 0
[16389]STUN trying 0
[16390]STUN trying 0
[16391]STUN trying 0
AUD:Stop PSTN Tone
[0]FM Alert Stop RxTx (c=0024eaa4;a=0)
[1:0]AUD Rel Call
[0]FM Alert Stop RxTx (c=0024900c;a=0)
[0:0]AUD Rel Call
CC:Ended
DLG Terminated 2cf2f8
[1:0]CC:STUN OK:c0a80196->7cbcf270, 5062->1567 16388->16388
AUD:Stop PSTN Tone
Calling:201259@au.voxalot.com:0
[1:0]AUD ALLOC CALL (port=16388)
[1:0]RTP Rx Up
DLG Terminated 2cf264
Sess Terminated
Sess Terminated
TP Parser error: 34
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
[0]FM Alert Stop RxTx (c=0024eaa4;a=0)
[1:0]AUD Rel Call
TP Parser error: 34

I've tried this with another Voxalot account I set up, but same result. Other SIP addresses seem to call OK.

Cheers
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Unread 01-05-2008, 04:46 AM   #29
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Default Fwd SPA3102 PST calls to Vox Voicemail {solved}

I finally got this to work by changing the following:
  1. Enabling STUN server (thanks for the tip Ozimarco)
  2. Setting Use DNS SRV on the PSTN tab to YES (this solves the GetServerAddrErr message appearing in syslog)
  3. On the SIP tab, setting Substitute VIA Addr to YES and Send Resp To SRC Port to YES
  4. In my Voxalot account, setting a Call Connection rule that anyone calling my account gets diverted to VM. I also have Symmetric NAT Handling set to No in my Voxalot account

The call routing behaviour I now have is that on Line 1 busy or no answer, my calls go th Voxalot VM. Also, because I have call waiting active on Line 1, the user has the option of picking up the call before it goes to Voxalot.

Note: all CFwd fields on User tab are blank.

Thanks to everyone who contributed to this thread.

Cheers
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Unread 03-20-2008, 12:20 PM   #30
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Just couple questions to Kenthurst35, just wondering which VSP you using? and I sought of have the same problem but mine is any private callers calling wont allow my VSP to forward the call to my mobile,. wonder if private calls foward ok on your setup?

cheerz

ps: anyway glad you've solved your problem :-)

Last edited by gotalkvoip; 03-20-2008 at 12:32 PM.
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