Spawn extension (ext-intercom... I'm lost

FreePBX 15.0.23

According to the user, they attempted to perform an attended transfer of a call. While they were talking to the extension they were going to transfer to, they put the call on hold again and then was not able to get back to that call.

Now, they can receive calls, but any button they press on their phone automatically dials *80205 with 205 being the extension they were transferring to previously.

We have restarted the PBX, and both extensions but still the same issue.

The Phone displays “warning: line 4 seems to be stuck, please check”
No channels show as open or active in Asterisk but we did a “channel request hangup all” anyway
A lot more logs available of course, but this seems to be the most relevant bit.

*8145[2022-03-31 14:40:03] VERBOSE[29292][C-00000029] app_stack.c: Spawn extension (from-internal, 80205, 1) exited non-zero on ‘PJSIP/205-00000072’

28146[2022-03-31 14:40:03] VERBOSE[29292][C-00000029] app_stack.c: PJSIP/205-00000072 Internal Gosub(autoanswer,s,1(Ring Answer,;answer-after=0)) complete GOSUB_RETVAL=

28147[2022-03-31 14:40:03] VERBOSE[29292][C-00000029] app_dial.c: Called PJSIP/205/sip:[email protected]:12160

28148[2022-03-31 14:40:03] VERBOSE[12419] netsock2.c: Using SIP RTP Audio TOS bits 184

28149[2022-03-31 14:40:03] VERBOSE[12419] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.

28150[2022-03-31 14:40:03] VERBOSE[12419] netsock2.c: Using SIP RTP Audio CoS mark 5

28151[2022-03-31 14:40:03] VERBOSE[29292][C-00000029] app_dial.c: Connected line update to PJSIP/201-00000070 prevented.

28152[2022-03-31 14:40:03] VERBOSE[29292][C-00000029] app_dial.c: PJSIP/205-00000072 is ringing

28153[2022-03-31 14:40:04] VERBOSE[29292][C-00000029] app_dial.c: PJSIP/205-00000072 answered PJSIP/201-00000070

28154[2022-03-31 14:40:04] VERBOSE[29292][C-00000029] file.c: <PJSIP/205-00000072> Playing ‘beep.slin16’ (language ‘en’)

28155[2022-03-31 14:40:04] VERBOSE[29322][C-00000029] bridge_channel.c: Channel PJSIP/205-00000072 joined ‘simple_bridge’ basic-bridge <6c68e9e5-d872-4f21-a1c7-946c07c88e9d>

28156[2022-03-31 14:40:04] VERBOSE[29292][C-00000029] bridge_channel.c: Channel PJSIP/201-00000070 joined ‘simple_bridge’ basic-bridge <6c68e9e5-d872-4f21-a1c7-946c07c88e9d>

28157[2022-03-31 14:40:06] VERBOSE[29292][C-00000029] bridge_channel.c: Channel PJSIP/201-00000070 left ‘simple_bridge’ basic-bridge <6c68e9e5-d872-4f21-a1c7-946c07c88e9d>

28158[2022-03-31 14:40:06] VERBOSE[29322][C-00000029] bridge_channel.c: Channel PJSIP/205-00000072 left ‘simple_bridge’ basic-bridge <6c68e9e5-d872-4f21-a1c7-946c07c88e9d>

*28159[2022-03-31 14:40:06] VERBOSE[29292][C-00000029] pbx.c: Spawn extension (ext-intercom, 80205, 37) exited non-zero on ‘PJSIP/201-00000070’

Any help to get this back to normal for this extension would be greatly appreciated.

EDIT: This is the start of the log when any button is pressed on the phone

[2022-03-31 16:04:49] VERBOSE[2643] netsock2.c: Using SIP RTP Audio TOS bits 184
[2022-03-31 16:04:49] VERBOSE[2643] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2022-03-31 16:04:49] VERBOSE[2643] netsock2.c: Using SIP RTP Audio CoS mark 5
*[2022-03-31 16:04:49] VERBOSE[9414][C-00000001] pbx.c: Executing [*[email protected]:1] Goto(“PJSIP/201-00000000”, "ext-intercom,80205,1") in new stack
*[2022-03-31 16:04:49] VERBOSE[9414][C-00000001] pbx_builtins.c: Goto (ext-intercom,80205,1)
*[2022-03-31 16:04:49] VERBOSE[9414][C-00000001] pbx.c: Executing [[email protected]:1] Macro(“PJSIP/201-00000000”, “user-callerid,”) in new stack
[2022-03-31 16:04:49] WARNING[9414][C-00000001] app_macro.c: Macro() is deprecated and will be removed from a future version of Asterisk.
[2022-03-31 16:04:49] WARNING[9414][C-00000001] app_macro.c: Dialplan should be updated to use Gosub instead.

RESOLVED

So, the message “warning: line 4 seems to be stuck, please check”

actually means a button is stuck, and if a button is stuck and you press anything else, it will just dial that stuck button. This is on a ClearlyIP 270v2

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.