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Problem with Asterisk server, when call from outside "GSM"


(Hunterman) #1

Dear Friends
My lab >>>
AM have GSM gateway device and connected with Asterisk server by sip trunk,
information of Asterisk server trunk:
Trunk Name: GSM
Outside:
type=peer
quality=yes
qualify=yes
host=x.x.x.x"private ip" for gsm gateway device

Incoming:

empty

I need to call from mobile number GSM to Asterisk server to ring directly to extension like 3000 in Asterisk server,
after applied configuration in GSM devices all was ok, because i tried to call from GSM to number of trunk to Asterisk server but extension no’t ringing, after that Am open ssh with Asterisk server and the result showing below:

[Feb 12 14:21:12] – Executing [3000@from-sip-external:1] NoOp("PJSIP/anonymous-00000014", "Received incoming SIP connection from unknown peer to 3000") in new stack

[Feb 12 14:21:12] – Executing [3000@from-sip-external:2] Set("PJSIP/anonymous-00000014", "DID=3000") in new stack

[Feb 12 14:21:12] – Executing [3000@from-sip-external:3] Goto("PJSIP/anonymous-00000014", "s,1") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,1)

[Feb 12 14:21:12] – Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00000014", "1?setlanguage:checkanon") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,2)

[Feb 12 14:21:12] – Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00000014", "CHANNEL(language)=en") in new stack

[Feb 12 14:21:12] – Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00000014", "1?noanonymous") in new stack

[Feb 12 14:21:12] – Goto (from-sip-external,s,5)

[Feb 12 14:21:12] – Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00000014", "TIMEOUT(absolute)=15") in new stack

[Feb 12 14:21:12] – Channel will hangup at 2019-02-12 14:21:27.691 +03.

[2019-02-12 14:21:12] WARNING[10608][C-0000004b]: func_channel.c:463 func_channel_read: Unknown or unavailable item requested: ‘recvip’

[Feb 12 14:21:12] – Executing [s@from-sip-external:6] Log("PJSIP/anonymous-00000014", "WARNING,"Rejecting unknown SIP connection from "") in new stack

[2019-02-12 14:21:12] WARNING[10608][C-0000004b]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "

[Feb 12 14:21:12] – Executing [s@from-sip-external:7] Answer("PJSIP/anonymous-00000014", "") in new stack

[Feb 12 14:21:12] > 0x7f63fc008380 – Strict RTP learning after remote address set to: 192.168.33.169:8012

[Feb 12 14:21:13] > 0x7f63fc008380 – Strict RTP switching to RTP target address 192.168.33.169:8012 as source

[Feb 12 14:21:13] – Executing [s@from-sip-external:8] Wait("PJSIP/anonymous-00000014", "2") in new stack

[2019-02-12 14:21:13] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)

[2019-02-12 14:21:13] ERROR[10608][C-0000004b]: channel.c:8073 ast_channel_start_silence_generator: Could not set write format to SLINEAR

[Feb 12 14:21:15] – Executing [s@from-sip-external:9] Playback("PJSIP/anonymous-00000014", "ss-noservice") in new stack

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin16|g722|alaw|ulaw) -> (g723)

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: file.c:1245 ast_streamfile: Unable to open ss-noservice (format (g723)): Function not implemented

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: app_playback.c:492 playback_exec: Playback failed on PJSIP/anonymous-00000014 for ss-noservice

[Feb 12 14:21:15] – Executing [s@from-sip-external:10] PlayTones("PJSIP/anonymous-00000014", "congestion") in new stack

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: channel.c:5600 set_format: Unable to find a codec translation path: (slin) -> (g723)

[2019-02-12 14:21:15] WARNING[10608][C-0000004b]: indications.c:140 playtones_alloc: Unable to set ‘PJSIP/anonymous-00000014’ to signed linear format (write)

[2019-02-12 14:21:15] NOTICE[10608][C-0000004b]: app_playtones.c:98 handle_playtones: Unable to start playtones

[Feb 12 14:21:15] == Spawn extension (from-sip-external, s, 10) exited non-zero on ‘PJSIP/anonymous-00000014’

[Feb 12 14:21:15] – Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00000014", "") in new stack

[Feb 12 14:21:15] == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000014’

AsteriskNOW*CLI>

Any help to solve the problem ?
THANKS


(Lorne Gaetz) #2

Add a line to the peer details like:

context=from-trunk

and then create an inbound route to direct the calls.


(Hunterman) #3

Thank you for your replay
where i change you mean in outgoing like

context=from-trunk
type=peer
quality=yes
qualify=yes
host=x.x.x.x"private ip" for gsm gateway device

and for incoming ? empty


(Dave Burgess) #4

Your call is coming in from an anonymous (unknown) server. You’ll need to add your ITSP host address in your Trunk to process the incoming calls from anywhere except your GSM Gateway.


(system) closed #5

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