A call to several numbers from the group does not go through. What could be the problem?

Question

Asterisk

Good day!
I ran into such a problem that after setting up a group call in Asterisk, the call does not go through to several numbers. With him experience is not enough, so I can not understand where to dig. Internal calls from and to problematic numbers go through normally, exactly the same as if you redirect an external call from another handset. The problem is that an external call falling on a group, judging by the logs, does not go to several numbers.

An external call goes to an external number, where it is routed to this

list of numbers :

8001
8003
8007
8080
8083
8085
8086
8110

After that, by timeout, if no one has taken it, it is redirected to mobile numbers indicated by the same group. There are no problems with this.

Incoming call log

[2020-10-28 09:26:45] VERBOSE[4232][C-0000058e] sig_pri.c: -- Accepting call from '96xxxxxxxx' to '221xxxx' on channel 0/6, span 1
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:1] Goto("DAHDI/i1/96xxxxxxxx-612", "from-trunk,221xxxx,1") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Goto (from-trunk,221xxxx,1)
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:1] Set("DAHDI/i1/96xxxxxxxx-612", "__FROM_DID=221xxxx") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:2] Gosub("DAHDI/i1/96xxxxxxxx-612", "app-blacklist-check,s,1()") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:1] GotoIf("DAHDI/i1/96xxxxxxxx-612", "0?blacklisted") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:2] Set("DAHDI/i1/96xxxxxxxx-612", "CALLED_BLACKLIST=1") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:3] Return("DAHDI/i1/96xxxxxxxx-612", "") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:3] Set("DAHDI/i1/96xxxxxxxx-612", "CDR(did)=221xxxx") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:4] ExecIf("DAHDI/i1/96xxxxxxxx-612", "1 ?Set(CALLERID(name)=96xxxxxxxx)") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:5] Set("DAHDI/i1/96xxxxxxxx-612", "CHANNEL(musicclass)=default") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:6] Set("DAHDI/i1/96xxxxxxxx-612", "__MOHCLASS=default") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:7] Set("DAHDI/i1/96xxxxxxxx-612", "__CALLINGPRES_SV=allowed") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:8] Set("DAHDI/i1/96xxxxxxxx-612", "CALLERPRES()=allowed_not_screened") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:9] Goto("DAHDI/i1/96xxxxxxxx-612", "temp-dest,s,1") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Goto (temp-dest,s,1)
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] pbx.c: -- Executing [[email protected]:1] Dial("DAHDI/i1/96xxxxxxxx-612", "SIP/8086&SIP/8085&SIP/8080&SIP/8007&SIP/8083&SIP/8088,20,tr") in new stack
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP TOS bits 184
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP TOS bits 184
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP TOS bits 184
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP TOS bits 184
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-10-28 09:26:45] WARNING[20014][C-0000058e] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP TOS bits 184
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] netsock2.c: == Using SIP RTP CoS mark 5
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- Called SIP/8086
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- Called SIP/8085
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- Called SIP/8080
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- Called SIP/8007
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- Called SIP/8088
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8088-0000132f connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8007-0000132e connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8080-0000132d connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8085-0000132c connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8086-0000132b connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8086-0000132b is ringing
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8080-0000132d is ringing
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8085-0000132c is ringing
[2020-10-28 09:26:45] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8088-0000132f is ringing
[2020-10-28 09:26:46] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8007-0000132e is ringing
[2020-10-28 09:26:47] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8007-0000132e connected line has changed. Saving it until answer for DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:26:47] VERBOSE[20014][C-0000058e] app_dial.c: -- SIP/8007-0000132e answered DAHDI/i1/96xxxxxxxx-612
[2020-10-28 09:29:32] VERBOSE[4232][C-0000058e] sig_pri.c: -- Span 1: Channel 0/6 got hangup request, cause 16

Based on the log, I can’t understand where the number 8088 comes from, because not listed anywhere. Plus, the call is not routed to the rest of the numbers from the group, which are indicated in the list above. I did dialplan show [email protected], I did not notice any problems in the output.

Output example

localhost*CLI> dialplan show [email protected]
[ Included context 'ext-local' created by 'pbx_config' ]
'8003' => hint: SIP/8003,CustomPresence:8003 [pbx_config]
1. Set(__RINGTIMER=${IF($[${DB(AMPUSER/8003/ringtimer)} > 0]?${DB(AMPUSER/8003/ringtimer)}:${RINGTIMER_DEFAULT})}) [pbx_config]
2. Macro(exten-vm,novm,8003,0,0,0) [pbx_config]
[dest] 3. Set(__PICKUPMARK=) [pbx_config]
4. Goto(${IVR_CONTEXT},return,1) [pbx_config]

[ Included context 'outrt-11' created by 'pbx_config' ]
'_XXXX' => 1. Macro(user-callerid,LIMIT,EXTERNAL,) [pbx_config]
2. Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})}) [pbx_config]
3. Set(_NODEST=) [pbx_config]
4. Gosub(sub-record-check,s,1(out,${EXTEN},)) [pbx_config]
5. Macro(dialout-trunk,4,${EXTEN},,off) [pbx_config]
6. Macro(outisbusy,) [pbx_config]

[ Included context 'bad-number' created by 'pbx_config' ]
'_X.' => 1. ResetCDR() [pbx_config]
2. NoCDR() [pbx_config]
3. Progress() [pbx_config]
4. Wait(1) [pbx_config]
5. Progress() [pbx_config]
6. Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer) [pbx_config]
7. Wait(1) [pbx_config]
8. Congestion(20) [pbx_config]
9. Hangup() [pbx_config]

-= 3 extensions (20 priorities) in 3 contexts. =-

Answer 1

Looks like the previous admin edited the config files.
cat /etc/asterisk/extensions_custom.conf
cat /etc/asterisk/extensions_override_freepbx.conf
There are two ways out
1. Correct the configs further, this piece fits your TK
[temp-dest]
exten => s,1,Dial(SIP/8086&SIP/ 8085&SIP/8080&SIP/8007&SIP/8083&SIP/8088,20,tr)
same => 2,Dial(DAHDI/g1/89xxxxxxxxx&DAHDI/g1/89xxxxxxxxx&DAHDI/g1/89xxxxxxxxx&DAHDI/g1/89xxxxxxxxx,30,tr)
same => 3,Dial( DAHDI/g1/89xxxxxxxxx&DAHDI/g1/89xxxxxxxxx&DAHDI/g1/89xxxxxxxxx,20,tr)
same => 4,Goto(temp-dest,s,1)
2. Configure normally via WEB