wassy83
(silvered.dragon)
April 3, 2018, 4:55pm
1
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
Stewart1
(Stewart)
April 4, 2018, 7:28am
2
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.
wassy83
(silvered.dragon)
April 4, 2018, 8:20am
3
ok I will try and posts the results
wassy83
(silvered.dragon)
April 4, 2018, 8:40am
4
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 my colleagues are really confused
wassy83
(silvered.dragon)
April 4, 2018, 1:44pm
5
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.
wassy83
(silvered.dragon)
April 4, 2018, 2:25pm
7
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
system
(system)
Closed
April 4, 2019, 2:25pm
8
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