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Kenthurst35 · 01-03-2008, 12:17 PM

Morning all

I have a voxalot account set up to call forward any inbound provider or number to voicemail. When I make a SIP call to this account using Gizmo, all seems to work as expected and the call goes straight to VM.

When I make the same SIP call from my ATA (Linksys SPA3102) as a result of an inbound PSTN call being picked up by the PSTN-To-VoIP gateway, the ATA never connects the call with VM (PSTN caller cannot hear any VM greeting) although some time after I get an email from Voxalot to say that a message has been left. Here is are the relevant syslog entries:

FXO:Start CNDD
Calling:201259@127.0.0.1:5060
[1:0]AUD ALLOC CALL (port=16408)
[1:0]RTP Rx Up
[0:0]AUD ALLOC CALL (port=16410)
[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=
RSE_DEBUG: reference domain:au.voxalot.com
FXO:PSTN Disconnect Tone
AUD:Stop PSTN Tone
FXO:On Hook
AUD:Stop PSTN Tone
FXO:Stop CNDD
[0]FM Alert Stop RxTx (c=0023f2a0;a=0)
[1:0]AUD Rel Call
[0]FM Alert Stop RxTx (c=0023a2d0;a=0)
[0:0]AUD Rel Call
CC:Ended
DLG Terminated 2b6088
DLG Terminated 2b6528
Sess Terminated
Sess Terminated
RSE_DEBUG: unref domain, au.voxalot.com
RSE_DEBUG: last unref for domain au.voxalot.com
AUD:Stop PSTN Tone
AUD:Stop PSTN Tone
Calling:201259@au.voxalot.com:0
[1:0]AUD ALLOC CALL (port=16412)
[1:0]RTP Rx Up
RSE_DEBUG: reference domain:au.voxalot.com
RSE_DEBUG: reference domain:au.voxalot.com
TP Parser error: 34
RSE_DEBUG: unref domain, au.voxalot.com
RSE_DEBUG: unref domain, au.voxalot.com
RSE_DEBUG: last unref for domain au.voxalot.com
RSE_DEBUG: reference domain:au.voxalot.com
RSE_DEBUG: unref domain, au.voxalot.com
RSE_DEBUG: last unref for domain au.voxalot.com
[0]FM Alert Stop RxTx (c=0023f2a0;a=0)
[1:0]AUD Rel Call
CC:Failed w/ Calling
AUD:Stop PSTN Tone
Sess Terminated

Funny thing is that i have tested other SIP calls from the gateway and they seem to ring and connect OK, so why is there a difference when calling Voxalot?

01-03-2008, 12:17 PM	#1
Kenthurst35 Junior Member Join Date: Dec 2007 Posts: 26 Thanks: 0 Thanked 1 Times in 1 Posts	SIP call to voxalot number Morning all I have a voxalot account set up to call forward any inbound provider or number to voicemail. When I make a SIP call to this account using Gizmo, all seems to work as expected and the call goes straight to VM. When I make the same SIP call from my ATA (Linksys SPA3102) as a result of an inbound PSTN call being picked up by the PSTN-To-VoIP gateway, the ATA never connects the call with VM (PSTN caller cannot hear any VM greeting) although some time after I get an email from Voxalot to say that a message has been left. Here is are the relevant syslog entries: FXO:Start CNDD Calling:201259@127.0.0.1:5060 [1:0]AUD ALLOC CALL (port=16408) [1:0]RTP Rx Up [0:0]AUD ALLOC CALL (port=16410) [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= RSE_DEBUG: reference domain:au.voxalot.com FXO:PSTN Disconnect Tone AUD:Stop PSTN Tone FXO:On Hook AUD:Stop PSTN Tone FXO:Stop CNDD [0]FM Alert Stop RxTx (c=0023f2a0;a=0) [1:0]AUD Rel Call [0]FM Alert Stop RxTx (c=0023a2d0;a=0) [0:0]AUD Rel Call CC:Ended DLG Terminated 2b6088 DLG Terminated 2b6528 Sess Terminated Sess Terminated RSE_DEBUG: unref domain, au.voxalot.com RSE_DEBUG: last unref for domain au.voxalot.com AUD:Stop PSTN Tone AUD:Stop PSTN Tone Calling:201259@au.voxalot.com:0 [1:0]AUD ALLOC CALL (port=16412) [1:0]RTP Rx Up RSE_DEBUG: reference domain:au.voxalot.com RSE_DEBUG: reference domain:au.voxalot.com TP Parser error: 34 RSE_DEBUG: unref domain, au.voxalot.com RSE_DEBUG: unref domain, au.voxalot.com RSE_DEBUG: last unref for domain au.voxalot.com RSE_DEBUG: reference domain:au.voxalot.com RSE_DEBUG: unref domain, au.voxalot.com RSE_DEBUG: last unref for domain au.voxalot.com [0]FM Alert Stop RxTx (c=0023f2a0;a=0) [1:0]AUD Rel Call CC:Failed w/ Calling AUD:Stop PSTN Tone Sess Terminated Funny thing is that i have tested other SIP calls from the gateway and they seem to ring and connect OK, so why is there a difference when calling Voxalot?