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Full Version: T2xP looses registration when call timeout
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I noticed that when you make a call and the phone does not get a response from the INVITE, it will display "call timeout" (which is okay) but then the phone loose its registration until the registration process kicks back in.

My customer also reported that when the system rings two phones (using the group feature of Freeswitch) and one picks the call, the other one looses registration.

Any idea what could go wrong?

Thanks
Benoit
Hi braymond,

Can you tell me the version of your T2X? We will release a new version of v71 made for freeswitch.
I am using the latest ones like 2.71.0.140 for the T28P
Hi braymond,

Do you use Cudatel PBX? We will release a version for freeswitch until next week.
Hi,

I am using a pure Freeswitch build, not an out-of-the-box bundle.

Just as a reference:
freeswitch@internal> version
FreeSWITCH Version 1.5.8b+git~20131127T184035Z~3e00d270d7~64bit (git 3e00d27 2013-11-27 18:40:35Z 64bit)

Thanks
Hi braymond,

Can you supply Syslog level 6, config.bin and pcap for me?
Before you export the syslog, please set log level as 6, and reboot the phone, then click Start,and reproduce the issue, then click Stop,and export the trace, syslog, config,bin to us.
(About where to export these files, please refer to attached screenshot.)
These three files are very important for us, hope you can kindly understand.
Aloha Braymond,

Did you ever solve the issue? We are having the same issue with a T41 phone that has 29.72.0.1 loaded. We will hear ringback, the display will show 'call timeout' then need to reregister.

Linksys devices on the same Asterisk system do not exhibit the behavior. Linksys devices are able to complete calls to the same number.

Attached is a SIP debug of the failed call.

Aloha,
Matt
Hi Matt,

No I did not solve the issue and have not really pursued it as it was fairly intermittent since my phones are directly connected on the same network as the PBX.

But your test shows exactly the real issue where the phone will force a REGISTER if its dialog (the INVITE in most cases) can't be fully completed.

Your problem looks to be NAT related. When DAHDI is answered, you can see Asterisk sending the 200 OK several times which is probably not even reaching the phone. So the phone has a pending INVITE not completed and then eventually gets the BYE from Asterisk when the call actually fails to be answered. The phone responds with a 481 and probably (my guess here) displays the call timeout and initiate a REGISTER.

It doesn't look like this phone should be anywhere other than on the same network.
The ironic thing is that the Asterisk server and IP phones are on the same subnet. Don't ask why they are using public IP's on their LAN.

By default Elastix has the 'NAT' setting to yes. We have tried with 'NAT' set to no and it did not solve the issue.

We have installed hundreds of yealink phones one multiple Elsatix systems and have never run into this issue before....it is almost like losing the registration causes the time out.

Waiting for Yealink support to respond.
Hi mattdarnell,

Sorry for the late reply.
Can you also supply the SIP file of Linksys?
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