05-26-2010 09:18 AM - edited 03-21-2019 09:26 AM
Here is our problem with the SPA2102. Please take a look at it. If you have a workable solution let me know.
When I make an outbound call from the Linksys SPA2102 to any number the any number has the call dropped after the call has been connected for roughly 23 seconds. From the phone Linksys SPA2102 the call thinks it is still going and doesn't seem to end.
I have included my conversation regarding the issue with our Metaswitch CSE. I have also included a wireshark capture of the call from between the SBG901 and the SPA2102 and a call trace of a dropped call from the Metaswitch. Firmware used in the test is 5.2.10, though the SPA2102s in the field are all 5.1.12 which is what is certified with our version of the Metaswitch. The Motorola modem in this case is running the most current version of firmware available from Motorola.
From our Metaswitch is sent a re-invite, which is done every 30 seconds from the initial invite (to verify the call is up and still connected), and the device sends us a 481 Does not exist, so the switch kills the call.
Here is the trace from our switch: 17:18:56.733 49341584 20122047:0001 SIP >> INVITE - DN: 6084934158 Call: 0 Trx: ce4f70 17:18:56.733 49341584 20122047:0001 SIP Content-Length: 164 17:18:56.733 49341584 20122047:0001 SIP Content-Type: application/sdp 17:18:56.733 49341584 20122047:0001 SIP v=0 17:18:56.733 49341584 20122047:0001 SIP o=- 189354392 189354392 IN IP4 209.83.10.25 17:18:56.733 49341584 20122047:0001 SIP s=- 17:18:56.733 49341584 20122047:0001 SIP c=IN IP4 209.83.10.25 17:18:56.733 49341584 20122047:0001 SIP t=0 0 17:18:56.733 49341584 20122047:0001 SIP m=audio 33884 RTP/AVP 18 0 101 17:18:56.733 49341584 20122047:0001 SIP a=rtpmap:101 telephone-event/8000 17:18:56.733 49341584 20122047:0001 SIP a=ptime:20 17:18:56.763 49341584 20122047:0001 SIP << 481 Call Leg/Transaction Does Not Exist - DN: 6084934158 Call: 0 Trx: ce4f70 17:18:56.763 49341584 20122047:0001 SIP To: 6084934158<sip:6084934158@sbc1.merr.com>;tag=fb92de4d 17:18:56.763 49341584 20122047:0001 SIP From: <sip:3935285@sbc1.merr.com>;tag=ms.merr.com+1+448163+b915a9 17:18:56.763 49341584 20122047:0001 SIP Via: SIP/2.0/UDP ms.merr.com;branch=z9hG4bK-b18e498378cfe8cf06cade1d17f2f5e7-ms 17:18:56.763 49341584 20122047:0001 SIP .merr.com-1 17:18:56.763 49341584 20122047:0001 SIP Call-ID: 0c0af54a005c49a4622ae6832b66e52d 17:18:56.763 49341584 20122047:0001 SIP CSeq: 215713198 INVITE 17:18:56.763 49341584 20122047:0001 SIP Contact: 6084934158<sip:CjuT8RTwoFyi-_KFSyswI8mG5j2tFJOOKYY6E9sdPX8gxJmSnk@209. 17:18:56.763 49341584 20122047:0001 SIP 83.10.45:5060> 17:18:56.763 49341584 20122047:0001 SIP Server: Linksys/SPA2102-5.2.10 (000e081c07ef) 17:18:56.763 49341584 20122047:0001 SIP Content-Length: 0
05-26-2010 09:30 AM
Dear Sir;
Can you send the configuration of the SPA2102? I'm interested to see the RTP Packet Size parameter on tab (provide this value instead of the whole config file please). If it is set to 0.030, please change it to 0.020 and apply changes (unit should reboot). Try again the call. May be a mismatch of packetization time.
Regards
Alberto
05-26-2010 03:21 PM
That is pretty much what we have configured, plus of course line information, codec, and a couple of other things. I did try manually changing the RTP Packet Size to what you recommended, but it was the same problem. The ATA itself doesn't realize that the call dropped and just hangs in silence.
05-26-2010 04:19 PM
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