Blind Transfer Problem with 12.5

[alaskatechnical]

We have several systems in the field and we have just recently installed 2 new 12.5 systems (The rest are 12.0 and work just fine). The problem we are having is when we do blind transfer the call never goes to voicemail (Well, it does sometimes after 3 minutes... Yes I timed it... but not everytime).

When the calling phone attempts the blind transfer it says Transferring and the called phone is ringing, after 2 minutes or so the calling phone stops ringing and it appears the call has been transferred. The called phone continues to ring for another minute and either goes to voicemail (I hear the voicemail prompt) or it just stops ringing and I (The caller) stays listening to hold music forever.

It doesn't seem to be handset dependent as I first thought (We first discovered the issue on a system with many Cisco 7941's) but we then discovered it happening on a system with only Yealink T26P or T28P. We have a almost identical system to the T26P/T28P system running 12.0 and it has no blind transfer problem.

I have NOT enabled the new Blind Transfer Call Back feature of 12.5.

Transfers work if we direct to voicemail transfer OR it's transferred from the Digital Receptionist.

I've been looking for solutions to the problem, so yes I have tried changing Codecs, Turning on and off "Supports Re-Invite" and "Support "Replaces" header" options, and the "I want to be able to accept more than 1 call at the same time".

Help!

 

[lneblett]

It would be highly helpful to post a log of the issue.

The ring is a SIP message and not an actual ring back like that of an analog phone. On the surface, it sounds like a network issue and by this I mean only that messages are not making it to and from their respective points successfully. Dual NIC card, wrong setting in phones, remote phones, VLANs, etc.

I am basing the observation on your explanation of the ring occurring for a number of minutes when I guess that you do not have the ring timeout set for this long and it should have done something else - vmail, forward, hangup, etc. in the meantime.

I have a number of 12.5 installs out there and none have this issue. If a common issue with 3CX, the board would be inaccessible due to the number of complaints.

Can you provide more details like the network setup and the aforementioned log?

 

[alaskatechnical]

Here is the snippet from the log:

06-May-2015 10:41:27.318 [CM503008]: Call(C:2600): Call is terminated
06-May-2015 10:41:27.316 Leg L:2600.1[Line:90005<<INCOMING-PSTN] is terminated: Cause: BYE from 10.0.100.14:5060
06-May-2015 10:38:14.821 Leg L:2601.1[Extn] is terminated: Cause: BYE from PBX
06-May-2015 10:38:14.816 [CM503020]: Call(C:2601): Normal call termination. Call originator: Extn:301. Reason: Terminated
06-May-2015 10:38:14.816 L:2601.1[Extn] failed to reach Extn:300, reason User Requested
06-May-2015 10:38:14.816 Call to T:Extn:300@[Dev:sip:[email protected]:5062] from L:2601.1[Extn] failed, cause: Cause: BYE from 10.0.100.91:5062
06-May-2015 10:38:14.816 Leg L:2601.2[Extn] is terminated: Cause: BYE from 10.0.100.91:5062
06-May-2015 10:38:14.611 [CM503025]: Call(C:2600): Calling T:Extn:300@[Dev:sip:[email protected]:5062] for L:2600.1[Line:90005<<INCOMING-PSTN]
06-May-2015 10:38:11.464 [CM503025]: Call(C:2601): Calling T:Extn:300@[Dev:sip:[email protected]:5062] for L:2601.1[Extn]
06-May-2015 10:38:11.413 [CM503027]: Call(C:2601): From: Extn:301 ("User 1" <sip:[email protected]:5060>) to T:Extn:300@[Dev:sip:[email protected]:5062]
06-May-2015 10:38:11.413 [CM503004]: Call(C:2601): Route 1: from L:2601.1[Extn] to T:Extn:300@[Dev:sip:[email protected]:5062]
06-May-2015 10:38:11.412 [CM503001]: Call(C:2601): Incoming call from Extn:301 to <sip:[email protected]:5060>
06-May-2015 10:38:08.243 Currently active calls - 2: [2598,2600]
06-May-2015 10:38:01.177 [CM503007]: Call(C:2600): Extn:301 has joined, contact <sip:[email protected]:5062>
06-May-2015 10:38:01.177 [CM503007]: Call(C:2600): Line:90005<<OUTSIDE-PSTN has joined, contact <sip:[email protected]:5060>
06-May-2015 10:38:01.175 L:2600.2[Extn] has joined to L:2600.1[Line:90005<<OUTSIDE-PSTN]
06-May-2015 10:37:58.718 [CM503025]: Call(C:2600): Calling T:Extn:301@[Dev:sip:[email protected]:5062] for L:2600.1[Line:90005<<OUTSIDE-PSTN]
06-May-2015 10:37:58.673 [CM503027]: Call(C:2600): From: Line:90005<<OUTSIDE-PSTN ("User 1:OUTSIDE CALLER " <sip:[email protected]:5060>) to T:Extn:301@[Dev:sip:[email protected]:5062]
06-May-2015 10:37:58.673 [CM503004]: Call(C:2600): Route 1: from L:2600.1[Line:90005<<INCOMING-PSTN] to T:Extn:301@[Dev:sip:[email protected]:5062]
06-May-2015 10:37:58.671 [CM503001]: Call(C:2600): Incoming call from Line:90005<<INCOMING-PSTN to <sip:[email protected]:5060>
06-May-2015 10:37:58.670 Line limit check: Current # of calls for line Lc:90005(@SN4970[<sip:[email protected]:5060>]) is 1; limit is 23
06-May-2015 10:37:58.665 [CM503012]: Inbound any hours rule (User 1) for 90005 forwards to DN:301

Does this shed any light on the issue? The phone rings as expected and if I answer the transfer the call transfers fine....

 

[lneblett]

Sorry, but I do not understand.

The initial post indicated that an incoming call could not be blind transferred to another extension.....that the called phone would ring for 2 minutes or more and might or might not ultimately go to voice mail.

I assume the following is the issue -
06-May-2015 10:38:14.611 [CM503025]: Call(C:2600): Calling T:Extn:300@[Dev:sip:[email protected]:5062] for L:2600.1[Line:90005<<INCOMING-PSTN]
06-May-2015 10:38:11.464 [CM503025]: Call(C:2601): Calling T:Extn:300@[Dev:sip:[email protected]:5062] for L:2601.1[Extn]

There is a 3 minute GAP between call 2601 and call 2600 (original).
Given the PSTN label, I assume this is a call coming in on a gateway? What make and model?

I am guessing that a wireshark will really be needed so as to see the interaction between all devices in the chain.

You might consider downloading and installing the softphone onto an iPhone or Android and making a SIP call in so as to bypass the PSTN and see what happens. I don't think it is an extension problem, but may have to do with the gateway.

 

[alaskatechnical]

It doesn't matter if I do a blind transfer from an internal call or an external call. The end result is the same. The called phone never goes to voicemail.

 

[leejor]

have a look at the reply from Jasit, it involves Yealink sets.

http://www.3cx.com/forums/forwarded-calls-not-going-to-voicemail-40991.html