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SPA3102: call forwarding, Voip-to-Voip

Call forwarding from my DID (Line 1) to my gsm does not work.
I configure in "Cfwd No Ans Dest:" my gsmnumber: 003247464xxxx@gw1

In de logfiles the right number is called. After 20 secs. But there is immidately: "DLG Terminated"
The phone keeps ringing .
Why ?
02-27-2010 16:12:42 Local2.Debug 192.168.0.51 Sess Terminated
02-27-2010 16:12:42 Local2.Debug 192.168.0.51 DLG Terminated 2a159c
02-27-2010 16:12:42 Local2.Debug 192.168.0.51 Calling:003247464xxxx@+329367xxxx@sip.justvoip.com:0
02-27-2010 16:12:42 Local2.Debug 192.168.0.51 [0:0]AUD Rel Call
02-27-2010 16:12:42 Local3.Debug 192.168.0.51 [0]FM Alert Stop RxTx (c=0024e5e8;a=0)
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 [0:0]RTP Rx Up
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 [0:0]AUD ALLOC CALL (port=16438)
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 Sess Terminated
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 DLG Terminated 2a1508
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 Calling:003247464xxxx@+329367xxxx@sip.justvoip.com:0
02-27-2010 16:12:22 Local2.Debug 192.168.0.51 [0:0]AUD Rel Call
02-27-2010 16:12:22 Local3.Debug 192.168.0.51 [0]FM Alert Stop RxTx (c=0024e5e8;a=0)
02-27-2010 16:12:02 Local2.Debug 192.168.0.51 [0:0]RTP Rx Up
02-27-2010 16:12:02 Local2.Debug 192.168.0.51 [0:0]AUD ALLOC CALL (port=16436)

What can be the problem ?

When I call up my gsm trough the phoneport it works fine. The line 1 dialplan directs the call also via gw1.
So account info , etc is correct.
The "Calling:"-statement is the same in the log.
But "DLG Terminated" only comes after "On hook" -> OK !


02-27-2010 14:38:38 Local2.Debug 192.168.0.51 CC:Clean Up
02-27-2010 14:38:37 Local2.Debug 192.168.0.51 Sess Terminated
02-27-2010 14:38:12 Local3.Debug 192.168.0.51 RSE_DEBUG: last unref for domain sip.justvoip.com
02-27-2010 14:38:12 Local3.Debug 192.168.0.51 RSE_DEBUG: unref domain, sip.justvoip.com
02-27-2010 14:38:10 Local3.Debug 192.168.0.51 RSE_DEBUG: unref domain, sip.justvoip.com
02-27-2010 14:38:10 Local3.Debug 192.168.0.51 RSE_DEBUG: unref domain, sip.justvoip.com
02-27-2010 14:38:05 Local2.Debug 192.168.0.51 DLG Terminated 26c074
02-27-2010 14:38:05 Local3.Debug 192.168.0.51 RSE_DEBUG: reference domain:sip.justvoip.com
02-27-2010 14:38:05 Local2.Debug 192.168.0.51 [0:0]AUD Rel Call
02-27-2010 14:38:05 Local3.Debug 192.168.0.51 [0]FM Alert Stop RxTx (c=0023a2d0;a=0)
02-27-2010 14:38:05 Local3.Debug 192.168.0.51 [0]On Hook
02-27-2010 14:38:04 Local2.Debug 192.168.0.51 [0:0]LAT-- 6(2)
02-27-2010 14:37:56 Local2.Debug 192.168.0.51 [0:0]LAT-- 6(2)
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]DEC INIT 0
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]RTP Rx 1st PKT @16388(2)
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 CC:CallProgress
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]RTCP Tx Up
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]RTP Tx Up (pt=0->4d48a899:25412)
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]ENC INIT 0
02-27-2010 14:37:40 Local3.Debug 192.168.0.51 RSE_DEBUG: reference domain:sip.justvoip.com
02-27-2010 14:37:40 Local3.Debug 192.168.0.51 RSE_DEBUG: reference domain:sip.justvoip.com
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]RTP Rx Up
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 [0:0]AUD ALLOC CALL (port=16388)
02-27-2010 14:37:40 Local2.Debug 192.168.0.51 Calling:003247464xxxx@+329367xxxx@sip.justvoip.com:0

Has anyone ever succesfully initiated a Voip-to-voip transfer ?

Is there a secondary setting to be done ?

Firmware is 5.1.10(GW) = most recent

Thanks for your help.

Mark

5 REPLIES
Cisco Employee

Re: SPA3102: call forwarding, Voip-to-Voip

Dear Sir;

Can you please send the web configuration of the device? In addition, can you please take the traces again but set the sip debug option in Line 1 and PSTN line as well, to see the SIP messages.

Regards
Alberto

New Member

Re: SPA3102: call forwarding, Voip-to-Voip

Hello,

here some new tests.

My DID-number is now at sipgate.co.uk (0044),

GW1 is Rynga.com

Is there a better way to send the SPA's configuration then screenshots ?

Regards ,

Mark

Cisco Employee

Re: SPA3102: call forwarding, Voip-to-Voip

Sir;

Please do the following:

- Open the webpage, go to admin advanced mode and then save the html full page (.mht). This will do the trick, make sure you delete (without submit changes) the confidential info (user id and pwd).

Regards

Alberto

New Member

Re: SPA3102: call forwarding, Voip-to-Voip

Hello ,

You can find the configuration in the attached file.

Regards,

Mark

Cisco Employee

Re: SPA3102: call forwarding, Voip-to-Voip

Dear Sir;

Your configuration looks ok, I'm escalating this to engineering to see if this is an issue or there is something I've not seen. Stay tuned.

regards
Alberto

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