Problem with Asterisk server, when call from outside “GSM”


#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


#2

It looks like “recvip” is for chan_sip while pjsip uses “local_addr”, according to this doc:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL

I could be misinterpreting that doc, but that’s what I would look at first as it seems that “recvip” is what is throwing the error.