Not receiving busy tone when transfering

hi to all
Freepbx is 14.0.2.10
asterisk is 13.19.1
ipphones are fanvil X4G
chan_sip is completaly disabled
call waiting is disabled

1 - external call is ringing
2 - ext1 pickup the call
3 - ext1 press transfer–>ext2–>dial
4 - ext2 is busy
5 - ext1 will not receive any tone just “call failed” in the display.

instead if:
1 - ext1 calls ext2
2 - ext2 is busy
3 - ext1 will receive the busy tone and then will hang up

is there a way to receive a busy tone instead of nothing during call transfer?
many thanks

I don’t have these phones, but it smells like a phone bug to me. I would expect the SIP busy response sent to ext1 to be identical for the two situations. You can confirm this with pjsip set logger on and comparing the cases.

ok I will try and posts the results

This is what I got during transfer if destination is busy (no busy tone just error on display)

freepbx*CLI> pjsip show history where addr = 192.168.25.176:5872
No. Timestamp (Dir) Address SIP Message
===== ========== ============================== ===================================
00082 1522830216 * ==> 192.168.25.176:5872 INVITE sip:[email protected]:5872 SIP/2.0
00083 1522830216 * <== 192.168.25.176:5872 SIP/2.0 100 Trying
00088 1522830217 * <== 192.168.25.176:5872 SIP/2.0 180 Ringing
00096 1522830219 * <== 192.168.25.176:5872 SIP/2.0 200 OK
00097 1522830219 * ==> 192.168.25.176:5872 ACK sip:[email protected]:5872 SIP/2.0
00204 1522830232 * <== 192.168.25.176:5872 INVITE sip:[email protected]:5060;user=phone SIP/2.0
00205 1522830232 * ==> 192.168.25.176:5872 SIP/2.0 200 OK
00208 1522830232 * <== 192.168.25.176:5872 ACK sip:[email protected]:5060 SIP/2.0
00241 1522830236 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00242 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 401 Unauthorized
00243 1522830236 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0
00244 1522830236 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00245 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 100 Trying
00246 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 486 Busy Here
00247 1522830236 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0
00258 1522830239 * <== 192.168.25.176:5872 BYE sip:[email protected]:5060 SIP/2.0
00259 1522830239 * ==> 192.168.25.176:5872 SIP/2.0 200 OK

and this is during a normal call if destinations is busy (receiving right busy tone)

freepbx*CLI> pjsip show history where addr = 192.168.25.176:5872
No. Timestamp (Dir) Address SIP Message
===== ========== ============================== ===================================
00082 1522830216 * ==> 192.168.25.176:5872 INVITE sip:[email protected]:5872 SIP/2.0
00083 1522830216 * <== 192.168.25.176:5872 SIP/2.0 100 Trying
00088 1522830217 * <== 192.168.25.176:5872 SIP/2.0 180 Ringing
00096 1522830219 * <== 192.168.25.176:5872 SIP/2.0 200 OK
00097 1522830219 * ==> 192.168.25.176:5872 ACK sip:[email protected]:5872 SIP/2.0
00204 1522830232 * <== 192.168.25.176:5872 INVITE sip:[email protected]:5060;user=phone SIP/2.0
00205 1522830232 * ==> 192.168.25.176:5872 SIP/2.0 200 OK
00208 1522830232 * <== 192.168.25.176:5872 ACK sip:[email protected]:5060 SIP/2.0
00241 1522830236 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00242 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 401 Unauthorized
00243 1522830236 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0
00244 1522830236 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00245 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 100 Trying
00246 1522830236 * ==> 192.168.25.176:5872 SIP/2.0 486 Busy Here
00247 1522830236 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0
00258 1522830239 * <== 192.168.25.176:5872 BYE sip:[email protected]:5060 SIP/2.0
00259 1522830239 * ==> 192.168.25.176:5872 SIP/2.0 200 OK
00424 1522830752 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00425 1522830752 * ==> 192.168.25.176:5872 SIP/2.0 401 Unauthorized
00426 1522830752 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0
00427 1522830752 * <== 192.168.25.176:5872 INVITE sip:[email protected];user=phone SIP/2.0
00428 1522830752 * ==> 192.168.25.176:5872 SIP/2.0 100 Trying
00429 1522830752 * ==> 192.168.25.176:5872 SIP/2.0 486 Busy Here
00431 1522830752 * <== 192.168.25.176:5872 ACK sip:[email protected];user=phone SIP/2.0

so I think is a bug on the phone but how to fix this :frowning: my colleagues are really confused

here is a log trough a syslog server connected to the ipphone during the error

Asterisk is sending a busy message. This seems like phone issue. You would need to contact phone manufacture on this.

ok thank you, I have wrote to fanvil and I will update this if they found a solution… these fanvil phones are really filled of bugs… it wasn’t a good choice to buy so many phones from this brand

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