Call Transfer and ring, but wont pickup


(Acrotec) #1

HI guys

We are using version FreePBX 14.0.13.40 and having the following issues.

When we transfer a call, the destination phone rings, but wont connect when picked up.
the call is able to be picked up back on the source phone. We are unable to transfer to any extension.
All phone can dial out as per normal


#2

Is this something that was working and is now failing? If so, are you aware of something that is likely related (FreePBX update, OS software update, new device, etc.)? If it never worked, is this a new system, or just the first time attempting to use transfer?

What device(s) are you using? Are you using the transfer key / softkey on the device, or transferring with DTMF tones e.g. *2? Does the other method fail the same way?

Paste the Asterisk log for a failed call at pastebin.freepbx.org and post the link here.


(Acrotec) #3

Thank you for your help,
Please find a link below to the logs
https://pastebin.freepbx.org/view/c1bb4e75

We did move the phone system to another site, new subnet and new Internet service. The system can receive calls and make calls from all phones


#4

I don’t know whether the log was somehow filtered, or it didn’t cover the correct period, but I don’t see any transfer activity at all.

I assume that the call in question was at 01:36:08 from 0280603137, answered by ext. 100 and transfer to 107 was initiated at 01:36:20. Is that correct? But when the log ends at 01:37:19, 100 and 107 are apparently still talking.

What device is ext. 100? What buttons were pressed to initiate the attended transfer? To complete the transfer?

If may be better for you to make a test call for the specific purpose of demonstrating the problem, with everything done within one minute. Along with the log, please indicate, with timestamps, what actions were taken and what the parties heard.


(Dave Burgess) #5

Acrotec, in case you’re curious why - it is possible to initiate a transfer from the phone that may or may not involve the resources of the PBX. When this happens, all kinds of hinky things can happen.


#6

You are correct and I forgot about that. Obihai IP phones (and also ATAs) can be configured to bridge calls locally. Some other brands may do this. Also, with almost any IP phone, you can establish a three-way call (conference), which doesn’t involve the PBX, and the ‘transferor’ could then press mute or just remain silent.

Phones with a ‘transfer on conference hangup’ feature do perform a regular (SIP REFER) transfer when the user who established the three-way call hangs up.


(Acrotec) #7

Hey guys
Thank you for your help,

Please find the logs below:
https://pastebin.freepbx.org/view/350127c9

Cheers


#8

Thanks, but you can’t expect readers to review more than 3 hours(!) of logs looking for trouble.
Please identify a specific transfer that failed and the time it occurred. Include information about the devices, whether the transfer key or DTMF was used, what the parties heard, etc.

Even better, make a test call that demonstrates the problem. The resulting log should be less than one minute long. If you do it at a time when the system is idle, this will be the only call in the log and much easier to follow.


(Acrotec) #9

Hey Guys,

Sorry for the long logs, Please find the below logs taken at a quite time:
https://pastebin.freepbx.org/view/33d5cc92

The dial in number is 0280603137 answered by Ext 100 then transferred to Ext 107

Thank you for your help,

cheers
Joel


#10

Line 985:
[2020-11-12 22:16:01] NOTICE[11828] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/100-000001aa' for lack of RTP activity in 30 seconds
Normally, extension 100 would have signaled ‘hold’ on the incoming call by sending a re-invite telling Asterisk not to send it RTP. Asterisk knows that this means hold and would then allow 300 seconds of no RTP before disconnecting that leg. I don’t know why this didn’t happen, but posting a new log that includes a SIP trace will likely show the problem.

To do that, at the Asterisk command prompt, type
pjsip set logger on
then make your test call.

However, I assume that in this case the conversation between 100 and 107 lasted less than 30 seconds so the disconnect for lack of RTP should not have happened. Can you please confirm that? However, there is no evidence of 100 even attempting to complete the transfer so that’s a separate problem (though it may have the same cause).

Along with the new log, please post make, model and version of the device at extension 100, as well as what buttons were pressed to call 107 and what was pressed to attempt to complete the transfer.


(Acrotec) #11

Hi Guys

Thank you for your help,
Please see logs attached.
Phone details
Yealink T46S
fw 66.81.0.70
hw 66.0.0.128.0.0.0

There pressing the ‘Transfer button’…

Logs
https://pastebin.freepbx.org/view/8897b43b

Phone Sys Log ext 100
https://pastebin.freepbx.org/view/54a9e7b5

Phone Sys log Ext 107
https://pastebin.freepbx.org/view/82859619

thanks guys for your help

cheers


#12

Sorry, but the log has neither a SIP trace nor the complete sequence (it starts in the middle of 100 calling 107).

When you type
pjsip set logger on
at the Asterisk command prompt (not a shell prompt), you should see
PJSIP Logging enabled
if not, post details. Also, please note that a reload (or pressing Apply Config) will cancel pjsip logging.

After issuing the above command, call in from your mobile, ask to be transferred to 107, answer it, have 100 complete the transfer, confirm the failure and hang up. The entire process should take less than one minute.