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Solved Call Transfer (Blind/Attended) no longer working

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ndom91

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Hi All,

So we've got two types of phones deployed, Grandstream GXP1628 and GXP2140. At first we thought it was just a bug with the GXP2140's, but I can also confirm that we cannot successfully transfer a call with the GXP1628 nor with the web interface.

What happens is the following:

1) Call comes in to extension A
2) ext. A answers
3) ext A presses transfer and dials 2-digit code for ext B
4) ext B Answers, person A says "you've got so and so on the line, can I transfer?"
5) person A presses transfer, selects line 1 to join the two calls together and hangs up.

In the web interface it looks as though the other extension is on the call with the original caller, however neither the original caller nor the transfered-to extension can hear one another and the call gets dropped about 10 seconds later.

Anyone have any idea whats going on here?!

---

3CX Verson: 16.0.641
Debian Stretch (9) latest updates installed.
GXP1640's are on fw v1.0.4.132
GXP2140's are on fw v1.0.11.6
Deutsche Telekom NGN SIP
Accept Multiple Calls is activated on all extensions
 
Hi,

Try the following:


  • Answer on the 1628
  • Press the transfer hard button 1599749912356.png
  • Dial the other extension and press the green hard button 1599749981114.png
  • One the other person answers and is informed simply press the transfer hard button again
 
Hi John,

Maybe I described it badly, but that is the process we are going through to do attended transfers. I just tested it again and it works up until the last step, where I press the hard transfer button again.

The call seems to be joined between the original caller and the target extension, as my phone returns to the idle screen. But neither the original caller nor the target extension can hear eachother and then the call gets dropped on both sides approximately 10 seconds later.
 
Last edited:
Would it be possible to perform a few transfers using your webclient, so that we can verify if the PBX can at least function as expected?

We can then look at the phones once this is verified.
Also please mention whether you were using custom phone templates, or default ones provided by 3CX
 
Yeah I've tried transfers via the web client as well as mentioned in the OP. That also does not work for some reason :/.

We are using custom phone templates, but we've only changed the idle screen layout.

EDIT:

Transfers, both attended and blind, worked without problem up until about a week or two ago. I applied the latest update, 16.0.641, thats the only change I can think of. We also applied the update before that a day or two before. So maybe it was the previous update as well.
 
I tried to do a transfer again via the web client and checked the logs afterward. This is what I found. It begins at the bottom and goes up.

0012341234 is the original caller (fake replacement number obviously..)
60 is the extension he dialed
73 is the extension I'm trying to transfer to.

It seems to make the request to replace extension 60 with 73 (the transfer), but then shortly thereafter receives "488 Not Acceptable Here" from 60 to the initial caller (012341234). Could that be it?

Code: 
09/11/2020 1:23:57 PM - L:1385.1[Extn:60] got Terminated Send Req BYE from 0.0.0.0:0 tid=ab05987775e3730a [email protected]:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---ab05987775e3730a;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: "Nico Domino"<sip:[email protected]>;tag=869415563
From: <sip:[email protected]>;tag=11da7322
Call-ID: [email protected]
CSeq: 2 BYE
User-Agent: 3CXPhoneSystem 16.0.6.641 (641)
Content-Length: 0

09/11/2020 1:23:57 PM - Terminated from "Nico Domino"<sip:[email protected]>;tag=869415563 to <sip:[email protected]>;tag=11da7322; reason: LocalBye
09/11/2020 1:23:57 PM - L:1385.1[Extn:60] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=fa873006e42edf74 [email protected]:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---fa873006e42edf74;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: "Nico Domino"<sip:[email protected]>;tag=869415563
From: <sip:[email protected]>;tag=11da7322
Call-ID: [email protected]
CSeq: 2 BYE
Content-Length: 0

09/11/2020 1:23:57 PM - SendMsg from <sip:[email protected]>;tag=11da7322 to "Nico Domino"<sip:[email protected]>;tag=869415563
09/11/2020 1:23:57 PM - L:1385.2[Extn:73] got Terminated Recv Req BYE from 192.168.11.50:5060 tid=1656191618 Call-ID=hSCvd9KFrpLn7QFSW0DJJQ..:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.50:5060;branch=z9hG4bK1656191618;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: "Nico Domino" <sip:[email protected]:5060>;tag=a1cdde5a
From: <sip:[email protected]>;tag=2030537257
Call-ID: hSCvd9KFrpLn7QFSW0DJJQ..
CSeq: 2 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Supported: replaces, path, timer
User-Agent: Grandstream GXP1628 1.0.4.132 000b82d38778
Content-Length: 0

09/11/2020 1:23:57 PM - Terminated from <sip:[email protected]>;tag=2030537257 to "Nico Domino" <sip:[email protected]:5060>;tag=a1cdde5a; reason: RemoteBye
09/11/2020 1:23:57 PM - L:1385.2[Extn:73] Sending: OnSendResp Send 200/BYE from 0.0.0.0:0 tid=1656191618 Call-ID=hSCvd9KFrpLn7QFSW0DJJQ..:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.50:5060;branch=z9hG4bK1656191618;rport=5060
Contact: <sip:[email protected]:5060>
To: "Nico Domino"<sip:[email protected]:5060>;tag=a1cdde5a
From: <sip:[email protected]>;tag=2030537257
Call-ID: hSCvd9KFrpLn7QFSW0DJJQ..
CSeq: 2 BYE
Content-Length: 0

09/11/2020 1:23:57 PM - SendMsg from "Nico Domino" <sip:[email protected]:5060>;tag=a1cdde5a to <sip:[email protected]>;tag=2030537257
09/11/2020 1:23:57 PM - L:1384.2[Extn:60] got Terminated Send Req BYE from 0.0.0.0:0 tid=c8a8ec655e68c271 Call-ID=prrBYhYo-xVmY6GFns6s9g..:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---c8a8ec655e68c271;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>;tag=1027635853
From: "+49012341234" <sip:[email protected]:5060;nf=e>;tag=fdb5852d
Call-ID: prrBYhYo-xVmY6GFns6s9g..
CSeq: 3 BYE
Content-Length: 0

09/11/2020 1:23:57 PM - Terminated from <sip:[email protected]>;tag=1027635853 to "+49012341234" <sip:[email protected]:5060;nf=e>;tag=fdb5852d; reason: LocalBye
09/11/2020 1:23:57 PM - L:1384.2[Extn:60] Sending: OnSendReq Send Req BYE from 0.0.0.0:0 tid=03560e0811c6ae50 Call-ID=prrBYhYo-xVmY6GFns6s9g..:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---03560e0811c6ae50;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>;tag=1027635853
From: "+49012341234" <sip:[email protected]:5060;nf=e>;tag=fdb5852d
Call-ID: prrBYhYo-xVmY6GFns6s9g..
CSeq: 3 BYE
Content-Length: 0


And then everything just begins terminating..


09/11/2020 1:23:57 PM - SendMsg from "+49012341234" <sip:[email protected]:5060;nf=e>;tag=fdb5852d to <sip:[email protected]>;tag=1027635853

09/11/2020 1:23:57 PM - L:1384.2[Extn:60] Sending: OnSendResp Send 488/INVITE from 0.0.0.0:0 tid=194074414 Call-ID=prrBYhYo-xVmY6GFns6s9g..:
SIP/2.0 ***488 Not Acceptable Here***
Via: SIP/2.0/UDP 192.168.30.200:5060;branch=z9hG4bK194074414;rport=5060
To: "+49012341234"<sip:[email protected]:5060;nf=e>;tag=fdb5852d
From: <sip:[email protected]>;tag=1027635853
Call-ID: prrBYhYo-xVmY6GFns6s9g..
CSeq: 2 INVITE
Content-Length: 0

09/11/2020 1:23:57 PM - SendMsg from "+49012341234" <sip:[email protected]:5060;nf=e>;tag=fdb5852d to <sip:[email protected]>;tag=1027635853

09/11/2020 1:23:57 PM - Route to L:1384.3[Extn:73] sends Invite-OUT Send Req INVITE from 0.0.0.0:0 tid=64280c0f73ccf158 Call-ID=EquRocxe-qy9KUAhxj2RWw..:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---64280c0f73ccf158;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "+49012341234"<sip:[email protected]:5060;nf=e>;tag=4a9c1636
Call-ID: EquRocxe-qy9KUAhxj2RWw..
CSeq: 1 INVITE
Alert-Info: <http://www.notused.invalidtld>;info=external
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Supported: replaces, timer
Referred-By: <sip:[email protected]:5060>
Replaces: hSCvd9KFrpLn7QFSW0DJJQ..;to-tag=2030537257;from-tag=a1cdde5a
Content-Length: 0

09/11/2020 1:23:57 PM - Call(C:1384): Referred(i): src="+49012341234"<sip:[email protected]:5060;nf=e>; dst=<sip:[email protected]:5060>

09/11/2020 1:23:57 PM - request repalace=L:1384.2[Extn:60] With=L:1385.2[Extn:73];ReferTo=<sip:[email protected]>;ReferBy=<sip:[email protected]:5060>
 
Do ALL the extensions in question have these options enabled?
1599824812421.png
 
Yes, as well as PBX Delivers Audio in our case.
 
And is your trunk exactly like the picture below? Especially the circled part.

1599825455241.png
 
Btw with a blind transfer via web interface it works like this currently:

I hit blind transfer, the phone rings at the other extension. As soon as the transfered-to extension picks up, my extension hangs up, but the original caller just continually hears the on-hold music. It never connects him back with the transfered-to extension
 
Yes, please remove those two options. They will cause problems like the ones you described.
 
Glad to hear it was fixed :)

Note: those are disabled by default in our templates
 
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