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Any method can call from gsm mobile number to direct extension in freepbx like extension 100


(Hunterman) #1

Hello everyone

I need to make call from my mobile gsm to Asterisk server and go direct to extension like 100,
information about lab:

  • I can call from extension 100 to mobile number gsm (because Am applied all configure between GSM gateway device and Asterisk server by using trunk and outbound with “prefix and matched pattern like (prefix # /matched pattern 7)” .
  • Now Am need to how can make call from number mobile GSM to direct extension for exchange by using Asterisk server to ring extension like 100 ?

Any help?

THANKS


(Dave Burgess) #2

Set your inbound route to match the CID of your GSM phone and the call will direct wherever you tell it to go. Extension 100 is a reasonable extension number; direct to voicemail might be a problem, but getting calls to go there should be basic.


(Hunterman) #3

"Set your inbound route to match the CID of your GSM phone and the call will direct wherever you tell it to go. "
I tried this step but also not working
When i enter ssh to Asterisk server and write asterisk -vvvvvr to monitor the case, After call from gsm to asterisk server like international number I see with ssh,

== Setting global variable ‘SIPDOMAIN’ to ‘192.168.62.153’
– Executing [680@from-sip-external:1] NoOp(“PJSIP/anonymous-0000001e”, “Received incoming SIP connection from unknown peer to 680”) in new stack
– Executing [680@from-sip-external:2] Set(“PJSIP/anonymous-0000001e”, “DID=680”) in new stack
– Executing [680@from-sip-external:3] Goto(“PJSIP/anonymous-0000001e”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-0000001e”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-0000001e”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-0000001e”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-0000001e”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2019-02-13 13:23:30.116 +03.
[2019-02-13 13:23:15] WARNING[15389][C-00000056]: func_channel.c:463 func_channel_read: Unknown or unavailable item requested: ‘recvip’
– Executing [s@from-sip-external:6] Log(“PJSIP/anonymous-0000001e”, "WARNING,"Rejecting unknown SIP connection from “”) in new stack
[2019-02-13 13:23:15] WARNING[15389][C-00000056]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
– Executing [s@from-sip-external:7] Answer(“PJSIP/anonymous-0000001e”, “”) in new stack
> 0x7f6390827fd0 – Strict RTP learning after remote address set to: 192.168.33.169:8012
> 0x7f6390827fd0 – Strict RTP switching to RTP target address 192.168.33.169:8012 as source
– Executing [s@from-sip-external:8] Wait(“PJSIP/anonymous-0000001e”, “2”) in new stack
[2019-02-13 13:23:15] WARNING[15389][C-00000056]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)
[2019-02-13 13:23:15] ERROR[15389][C-00000056]: channel.c:8073 ast_channel_start_silence_generator: Could not set write format to SLINEAR
– Executing [s@from-sip-external:9] Playback(“PJSIP/anonymous-0000001e”, “ss-noservice”) in new stack
[2019-02-13 13:23:17] WARNING[15389][C-00000056]: channel.c:5600 set_format: Unable to find a codec translation path: (slin16|g722|alaw|ulaw) -> (g723)
[2019-02-13 13:23:17] WARNING[15389][C-00000056]: file.c:1245 ast_streamfile: Unable to open ss-noservice (format (g723)): Function not implemented
[2019-02-13 13:23:17] WARNING[15389][C-00000056]: app_playback.c:492 playback_exec: Playback failed on PJSIP/anonymous-0000001e for ss-noservice
– Executing [s@from-sip-external:10] PlayTones(“PJSIP/anonymous-0000001e”, “congestion”) in new stack
[2019-02-13 13:23:17] WARNING[15389][C-00000056]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)
[2019-02-13 13:23:17] WARNING[15389][C-00000056]: indications.c:140 playtones_alloc: Unable to set ‘PJSIP/anonymous-0000001e’ to signed linear format (write)
[2019-02-13 13:23:17] NOTICE[15389][C-00000056]: app_playtones.c:98 handle_playtones: Unable to start playtones
== Spawn extension (from-sip-external, s, 10) exited non-zero on ‘PJSIP/anonymous-0000001e’
– Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-0000001e”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-0000001e’

I think with Asterisk server can’t to route to direct extension.
Any help


(Dave Burgess) #4

That’s true. You need to set up an inbound route with a DID of 680.
Point the destination of the inbound route to extension 100.

See what happens.