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Good Day Any can help how to point DID to different IVR or Call Menu.
I Have a 10did i want to group in 3 . and put different call menu per group. may did defalut is set on call menu, but all the did read only the call menu set on the 1st group.. other did i configure out to different group already.. but didnt route to their resfective call menu.

Thank you

Looking at the Asterisk CLI output when you place calls to these DIDs, they they arrive at your system with different extensions or as the same?

Sir upon they arrive on same extension .. and same call menu.. i think all the config of my did is connected to defualt did? is that possible to dissable the default did.

Please provide the Asterisk CLI output for when a call arrives at both of these extensions.

== Using SIP RTP CoS mark 5
[Oct 3 13:14:32] -- Executing [8036@from-gsm11.4:1] AGI("SIP/8044-0000002f", "agi-DID_route.agi") in new stack
[Oct 3 13:14:32] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Oct 3 13:14:32] -- <SIP/8044-0000002f>AGI Script agi-DID_route.agi completed, returning 0
[Oct 3 13:14:32] -- Executing [99909*1***DID@default:1] Answer("SIP/8044-0000002f", "") in new stack
[Oct 3 13:14:32] > 0x7fa7740be8d0 -- Probation passed - setting RTP source address to 10.10.0.15:13002
[Oct 3 13:14:32] -- Executing [99909*1***DID@default:2] AGI("SIP/8044-0000002f", "agi-VDAD_ALL_inbound.agi") in new stack
[Oct 3 13:14:32] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Oct 3 13:14:33] -- <SIP/8044-0000002f> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:33] -- <SIP/8044-0000002f> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:33] -- <SIP/8044-0000002f> Playing 'mars888ty1.slin' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:36] -- <SIP/8044-0000002f> Playing 'monitor.slin' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:44] -- Started music on hold, class 'Mars888OnHoldMusic', on SIP/8044-0000002f
[Oct 3 13:14:47] -- Stopped music on hold on SIP/8044-0000002f
[Oct 3 13:14:47] -- <SIP/8044-0000002f> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:47] -- <SIP/8044-0000002f> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:47] -- <SIP/8044-0000002f> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:47] -- <SIP/8044-0000002f> Playing 'engage1.slin' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 3 13:14:57] -- <SIP/8044-0000002f>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[Oct 3 13:14:57] == Spawn extension (default, 99909*1***DID, 2) exited non-zero on 'SIP/8044-0000002f'
[Oct 3 13:14:57] -- Executing [h@default:1] AGI("SIP/8044-0000002f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 3 13:14:57] -- <SIP/8044-0000002f>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Do you have your carrier set to trunkinbound context?

Is "8036" the extension that calls come in on?

Have you tried adding 8036 as a DID in your system?

Sir The carrier context is GSM
8036 came to my gsm gateway.
yes

Sir This my other log on cli another number
= Using SIP RTP CoS mark 5
[Oct 7 08:29:45] -- Executing [09234020220@hgb8893:1] AGI("SIP/8893-000007d6", "agi-NVA_recording.agi,BOTH------Y---N---Y---N") in new stack
[Oct 7 08:29:45] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Oct 7 08:29:45] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171007082945_8893_09234020220)
[Oct 7 08:29:45] -- <SIP/8893-000007d6>AGI Script agi-NVA_recording.agi completed, returning 0
[Oct 7 08:29:45] -- Executing [09234020220@hgb8893:2] Goto("SIP/8893-000007d6", "default,09234020220,1") in new stack
[Oct 7 08:29:45] -- Goto (default,09234020220,1)
[Oct 7 08:29:45] -- Executing [09234020220@default:1] AGI("SIP/8893-000007d6", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 7 08:29:45] -- <SIP/8893-000007d6>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 7 08:29:45] -- Executing [09234020220@default:2] Dial("SIP/8893-000007d6", "sip/8023/09234020220") in new stack
[Oct 7 08:29:45] == Using SIP RTP CoS mark 5
[Oct 7 08:29:45] -- Called sip/8023/09234020220