BUG: 1 incoming call connected to another

[mattlandis]

trace.txt** call flow **

 

nexvortex siptrunk -> HuntGroup(extensions a b c) -> snom370

 

 

** How to reproduce issue **

 

extension A B and C are in a HuntgroupA (all snom 370's)

 

call#1 comes through HuntgroupA to extension A

 

extension A is now talking to Call#1

 

call#2 comes within 15seconds of Call#1 (important) is ringing HuntgroupA

 

while ringgroupA is ringing, if extension A presses tranfer button, call#1 and call#2 are connected.

 

 

snom ONE all one versions 3981 and up.

snom 370 - 8.4.32

 

see sip trace attatched.

[skyplonk]

** call flow **

 

nexvortex siptrunk -> HuntGroup(extensions a b c) -> snom370

 

 

** How to reproduce issue **

 

extension A B and C are in a HuntgroupA (all snom 370's)

 

call#1 comes through HuntgroupA to extension A

 

extension A is now talking to Call#1

 

call#2 is ringing HuntgroupA

 

while ringgroupA is ringing, if extension A presses tranfer button, call#1 and call#2 are connected.

 

 

snom ONE all one versions 3981 and up.

snom 370 - 8.4.32

 

see sip trace attatched.

 

Good work! we have had customers reporting this off and on for a while now, but I have never been able to recreate the issue. This sounds like it to me. Also seems to have the same issue with a mix of agent groups and hunt groups. The same thing happens if we make call 1 into a agent group..

[Vodia support]

I tested this scenario with 2011-4.5.0.1005 Alpha Monocerotids exactly as you mention above using 2 pbx. 1 Pbx has extension 100, 200. PBX 2 has 40,41,42 in the hunt group when the 2nd call comes, I press the transfer button, it does not conjoin the calls instead the phone waits for some input. The phones are using version 8.4.34

 

 

Here is link to the latest snomONE build.

 

http://wiki.snomone.com/index.php?title=Upgrades#snom_ONE.2C_Release_2011-4.3.0.5021.2C_11.2F10.2F2011

 

as usual back up your files.

[mattlandis]

what do you see in the sip trace that we provided?

 

it does it every time on the system in question so there is no question it is happening. ;-)

[Vodia PBX]

I guess Mr X wanted to ask you if you could back up your system, load the latest version and check if the problem was solved...

[mattlandis]

We actually had the latest version and that is when we noticed the issue first.

 

Then tried the next older version...and so on. No version we tried fixed it.

[Steve B]

We actually had the latest version and that is when we noticed the issue first.

 

Then tried the next older version...and so on. No version we tried fixed it.

 

I think they were referring to the Beta version that is out, did you try that one?

[pbx support]

We tested both v4.3 & v4.5 PBX versions with a mix of snom phones. We could not reproduce it here.

But need to double check if there is any issue on 370 with 8.4.32 versions.

[mattlandis]

we just installed a new snom one system and it does not do it on the new system so system specific.

 

But we did nothing tricky with this system, so i suspect we uncovered an esoteric bug. (like the call pickup one fixed in last release) But our customer is very ansy...any answers?

[pbx support]

Hmm... Are there any custom button profile that somehow interfering here? just guessing here..

[mattlandis]

we verified that the transfer button is TRANSFER.

 

every other senario transfer looks correct.

 

Have you taken a look at the logs we sent? What are you seeing?

 

(We really preferring not to just chat and kick around on this one)

 

Also, we've seen no response on ticket 20111222292.

 

thanks,

[Vodia support]

The ticket has been updated, awhile ago. We have uploaded your domain.tar file on our windows machine and downgraded our phone to 8.4.32 we tested you scenario on 4.2.1.4025 "pbxnsip" and we could not reproduce you scenario. If you can reproduce it, follow the direction on the ticket and update it with the new log file. Thanks

[andrewgroup]

This can happen, and it's important to ask your SIP provider for the RTP Port ranges they are expecting to be used. Be sure to match these port ranges on you PBX. In the case of Windows PBX's, Microsoft made a few updates in 2011 to battle security breached with DNS, and other services, and the OS will allocate a random range of ports during normal operations. Good reading begins here - http://support.microsoft.com/kb/956188

Based on our analysis, RTP streams were becoming crossed.