I cannot transfer calls between extensions. I have installed a new FreePBX 15.0.16.75 I have created 5 pjsip extensions, 10,11,12, 13, 14 and I can do everything but transfer extensions. Neither in blind nor attended mode, it is as if they are locked.
Can you help me? where can i check? Cheers
Show us an example from the /var/log/asterisk/full file and post it to https://pastebin.freepbx.org/
[2020-11-30 17:38:55] VERBOSE[18997][C-0000000d] app_dial.c: Connected line update to SIP/SIP-964891249-00000006 prevented.
[2020-11-30 17:38:55] VERBOSE[18997][C-0000000d] app_dial.c: PJSIP/13-00000011 is ringing
[2020-11-30 17:38:55] VERBOSE[18997][C-0000000d] app_dial.c: PJSIP/13-00000011 is ringing
[2020-11-30 17:38:57] VERBOSE[18997][C-0000000d] app_dial.c: PJSIP/13-00000011 answered SIP/SIP-964891249-00000006
[2020-11-30 17:38:57] VERBOSE[19003][C-0000000d] bridge_channel.c: Channel PJSIP/13-00000011 joined ‘simple_bridge’ basic-bridge <32dce16b-2289-481d-a3d5-42e546e76b09>
[2020-11-30 17:38:57] VERBOSE[18997][C-0000000d] bridge_channel.c: Channel SIP/SIP-964891249-00000006 joined ‘simple_bridge’ basic-bridge <32dce16b-2289-481d-a3d5-42e546e76b09>
[2020-11-30 17:39:27] NOTICE[1425] res_pjsip_sdp_rtp.c: Disconnecting channel ‘PJSIP/13-00000011’ for lack of audio RTP activity in 32 seconds
[2020-11-30 17:39:27] VERBOSE[19003][C-0000000d] bridge_channel.c: Channel PJSIP/13-00000011 left ‘simple_bridge’ basic-bridge <32dce16b-2289-481d-a3d5-42e546e76b09>
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] bridge_channel.c: Channel SIP/SIP-964891249-00000006 left ‘simple_bridge’ basic-bridge <32dce16b-2289-481d-a3d5-42e546e76b09>
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] app_macro.c: Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘SIP/SIP-964891249-00000006’ in macro ‘dial-one’
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] app_macro.c: Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘SIP/SIP-964891249-00000006’ in macro ‘exten-vm’
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Spawn extension (from-did-direct, 13, 3) exited non-zero on ‘SIP/SIP-964891249-00000006’
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [h@from-did-direct:1] Macro(“SIP/SIP-964891249-00000006”, “hangupcall,”) in new stack
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/SIP-964891249-00000006”, “1?theend”) in new stack
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/SIP-964891249-00000006”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“SIP/SIP-964891249-00000006”, "PJSIP/13-00000011 montior file= ") in new stack
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“SIP/SIP-964891249-00000006”, “1?skipagi”) in new stack
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2020-11-30 17:39:27] VERBOSE[18997][C-0000000d] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“SIP/SIP-964891249-00000006”, “”) in new stack
I try to transfer from extension 13 to 14 and it doesn’t work
Why I do if I post in the website you say?
and in console you can see this:
Goto (macro-exten-vm,s,13)
– Executing [s@macro-exten-vm:13] GosubIf(“PJSIP/14-00000012”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:14] Macro(“PJSIP/14-00000012”, “dial-one,HhTtr,13”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/14-00000012”, “DEXTEN=13”) in new stack
– Executing [s@macro-dial-one:2] ExecIf(“PJSIP/14-00000012”, “0?Set(__EXTTOCALL=13)”) in new stack
– Executing [s@macro-dial-one:3] Set(“PJSIP/14-00000012”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“PJSIP/14-00000012”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/14-00000012”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:6] GotoIf(“PJSIP/14-00000012”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,9)
– Executing [s@macro-dial-one:9] GotoIf(“PJSIP/14-00000012”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/14-00000012”, “0?continue”) in new stack
– Executing [s@macro-dial-one:11] Set(“PJSIP/14-00000012”, “EXTHASCW=”) in new stack
– Executing [s@macro-dial-one:12] GotoIf(“PJSIP/14-00000012”, “1?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,13)
– Executing [s@macro-dial-one:13] GotoIf(“PJSIP/14-00000012”, “0?docfu:skip3”) in new stack
– Goto (macro-dial-one,s,17)
– Executing [s@macro-dial-one:17] GotoIf(“PJSIP/14-00000012”, “1?next2:continue”) in new stack
– Goto (macro-dial-one,s,18)
– Executing [s@macro-dial-one:18] GotoIf(“PJSIP/14-00000012”, “1?continue”) in new stack
– Goto (macro-dial-one,s,26)
– Executing [s@macro-dial-one:26] GotoIf(“PJSIP/14-00000012”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:27] GosubIf(“PJSIP/14-00000012”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/14-00000012”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“PJSIP/14-00000012”, “DEVICES=13”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/14-00000012”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/14-00000012”, “0?Set(DEVICES=3)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“PJSIP/14-00000012”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“PJSIP/14-00000012”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“PJSIP/14-00000012”, “THISDIAL=PJSIP/13”) in new stack
– Executing [dstring@macro-dial-one:8] GotoIf(“PJSIP/14-00000012”, “0?docheck”) in new stack
– Executing [dstring@macro-dial-one:9] NoOp(“PJSIP/14-00000012”, “Debug: Found PJSIP Destination PJSIP/13”) in new stack
– Executing [dstring@macro-dial-one:10] GotoIf(“PJSIP/14-00000012”, “0?doset”) in new stack
– Executing [dstring@macro-dial-one:11] NoOp(“PJSIP/14-00000012”, “Debug: Updating PJSIP Destination with PJSIP_DIAL_CONTACTS”) in new stack
– Executing [dstring@macro-dial-one:12] Set(“PJSIP/14-00000012”, “THISDIAL=PJSIP/13/sip:[email protected]:5060”) in new stack
– Executing [dstring@macro-dial-one:13] ExecIf(“PJSIP/14-00000012”, “0?Set(DIALSTATUS=CHANUNAVAIL)”) in new stack
– Executing [dstring@macro-dial-one:14] GotoIf(“PJSIP/14-00000012”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:15] Set(“PJSIP/14-00000012”, “DSTRING=PJSIP/13/sip:[email protected]:5060&”) in new stack
– Executing [dstring@macro-dial-one:16] Set(“PJSIP/14-00000012”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:17] GotoIf(“PJSIP/14-00000012”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:18] ExecIf(“PJSIP/14-00000012”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:19] Set(“PJSIP/14-00000012”, “DSTRING=PJSIP/13/sip:[email protected]:5060”) in new stack
– Executing [dstring@macro-dial-one:20] Return(“PJSIP/14-00000012”, “”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“PJSIP/14-00000012”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/14-00000012”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:30] GosubIf(“PJSIP/14-00000012”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“PJSIP/14-00000012”, “DB(CALLTRACE/13)=14”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“PJSIP/14-00000012”, “”) in new stack
– Executing [s@macro-dial-one:31] Set(“PJSIP/14-00000012”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:32] GosubIf(“PJSIP/14-00000012”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:33] NoOp(“PJSIP/14-00000012”, "Blind Transfer: , Attended Transfer: , User: 14, Alert Info: ") in new stack
– Executing [s@macro-dial-one:34] ExecIf(“PJSIP/14-00000012”, “1?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/14-00000012”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/14-00000012”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/14-00000012”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/14-00000012”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] GosubIf(“PJSIP/14-00000012”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:40] ExecIf(“PJSIP/14-00000012”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:41] GosubIf(“PJSIP/14-00000012”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:42] Set(“PJSIP/14-00000012”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/14-00000012”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:44] GotoIf(“PJSIP/14-00000012”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/14-00000012”, “0?godial”) in new stack
– Executing [s@macro-dial-one:46] Gosub(“PJSIP/14-00000012”, “sub-presencestate-display,s,1(13)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“PJSIP/14-00000012”, “state-not_set,1”) in new stack
– Goto (sub-presencestate-display,state-not_set,1)
– Executing [state-not_set@sub-presencestate-display:1] Set(“PJSIP/14-00000012”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“PJSIP/14-00000012”, “”) in new stack
– Executing [s@macro-dial-one:47] Set(“PJSIP/14-00000012”, “CONNECTEDLINE(name,i)=prueba”) in new stack
– Executing [s@macro-dial-one:48] Set(“PJSIP/14-00000012”, “CONNECTEDLINE(num)=13”) in new stack
– Executing [s@macro-dial-one:49] Set(“PJSIP/14-00000012”, “D_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dial-one:50] Macro(“PJSIP/14-00000012”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“PJSIP/14-00000012”, “”) in new stack
– Executing [s@macro-dial-one:51] ExecIf(“PJSIP/14-00000012”, “0?Set(D_OPTIONS=HhtrI)”) in new stack
– Executing [s@macro-dial-one:52] ExecIf(“PJSIP/14-00000012”, “0?Set(CWRING=r(callwaiting)):Set(CWRING=)”) in new stack
– Executing [s@macro-dial-one:53] NoOp(“PJSIP/14-00000012”, “”) in new stack
– Executing [s@macro-dial-one:54] ExecIf(“PJSIP/14-00000012”, “0?Set(D_OPTIONS=HhTtrg)”) in new stack
– Executing [s@macro-dial-one:55] Dial(“PJSIP/14-00000012”, “PJSIP/13/sip:[email protected]:5060,HhTtrb(func-apply-sipheaders^s^1)”) in new stack
– PJSIP/13-00000013 Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/13-00000013”, “Applying SIP Headers to channel PJSIP/13-00000013”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/13-00000013”, “TECH=PJSIP”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“PJSIP/13-00000013”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/13-00000013”, “0”) in new stack
– Jumping to priority 12
– Executing [s@func-apply-sipheaders:13] Return(“PJSIP/13-00000013”, “”) in new stack
== Spawn extension (from-internal, 13, 1) exited non-zero on ‘PJSIP/13-00000013’
– PJSIP/13-00000013 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/13/sip:[email protected]:5060
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– PJSIP/13-00000013 is ringing
– PJSIP/13-00000013 is ringing
– PJSIP/13-00000013 answered PJSIP/14-00000012
> 0xb3b3af18 – Strict RTP learning after remote address set to: 192.168.0.226:5006
> 0xb3b7f0a0 – Strict RTP learning after remote address set to: 192.168.0.225:5004
– Channel PJSIP/13-00000013 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/14-00000012 joined ‘simple_bridge’ basic-bridge
[2020-11-30 17:54:58] NOTICE[1425]: res_pjsip_sdp_rtp.c:148 rtp_check_timeout: Disconnecting channel ‘PJSIP/13-00000013’ for lack of audio RTP activity in 39 seconds
– Channel PJSIP/13-00000013 left ‘simple_bridge’ basic-bridge
– Channel PJSIP/14-00000012 left ‘simple_bridge’ basic-bridge
== Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘PJSIP/14-00000012’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘PJSIP/14-00000012’ in macro ‘exten-vm’
== Spawn extension (from-internal, 13, 3) exited non-zero on ‘PJSIP/14-00000012’
– Executing [h@from-internal:1] Macro(“PJSIP/14-00000012”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/14-00000012”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
[2020-11-30 17:54:58] NOTICE[1425]: res_pjsip_sdp_rtp.c:148 rtp_check_timeout: Disconnecting channel ‘PJSIP/14-00000012’ for lack of audio RTP activity in 39 seconds
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/14-00000012”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/14-00000012”, "PJSIP/13-00000013 montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/14-00000012”, “1?skipagi”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] Hangup(“PJSIP/14-00000012”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/14-00000012’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/14-00000012’
– Channel PJSIP/13-00000013 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/14-00000012 joined ‘simple_bridge’ basic-bridge
**[2020-11-30 17:54:58] NOTICE[1425]: res_pjsip_sdp_rtp.c:148 rtp_check_timeout: Disconnecting channel ‘PJSIP/13-00000013’ for lack of audio RTP activity in 39 seconds**
– Channel PJSIP/13-00000013 left ‘simple_bridge’ basic-bridge
– Channel PJSIP/14-00000012 left ‘simple_bridge’ basic-bridge
This might be a NAT issue. The bridge is being created then disconnected.
Are all the devices including the FreePBX instance on the same network?
There is no nat, the phones are in the same subnet. the problem is that I cannot transfer calls it doesn’t work.I am using grandstream phones, could be a phone problem witch freepbx?
I use GrandStream Phones. I have upwards of 30 installations, all using GrandStream phones… Thats why I ask about the NAT…
What happens if you make the extensions longer? I use 4 digit extensions. Is it possible that when the transfer is dialed its trying to dial a different feature code?
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