Error in incoming calls, they come out as congested or Playing

When calling it gives me the message that the line is out of service, in the incoming routes it sends it to the IVR, to an extension or to a ring Groups, but everything gives me the same message.

In CDR reports i get is:

|Tue, 12 Apr 2022 14:29||1649773745.90|829XXXXXXX|||Playback|s [from-sip-external]|ANSWERED|00:06|||||
|Tue, 12 Apr 2022 14:16||1649772962.89|829XXXXXXX|||Playback|s [from-sip-external]|ANSWERED|00:20|||||
|Tue, 12 Apr 2022 14:05||1649772321.88|829XXXXXXX|||Congestion|s [from-sip-external]|ANSWERED|00:12|||||

And Asterisk log Files this:
22 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:1] NoOp(“PJSIP/anonymous-00000049”, “Received incoming SIP connection from unknown peer to 809XXXXXXX”) in new stack
23 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:2] Set(“PJSIP/anonymous-00000049”, “DID=809XXXXXXX”) in new stack
24 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:3] Goto(“PJSIP/anonymous-00000049”, “s,1”) in new stack
25 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx_builtins.c: Goto (from-sip-external,s,1)
26 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:1] GotoIf(“PJSIP/anonymous-00000049”, “1?setlanguage:checkanon”) in new stack
27 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx_builtins.c: Goto (from-sip-external,s,2)
28 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:2] Set(“PJSIP/anonymous-00000049”, “CHANNEL(language)=es”) in new stack
29 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:3] GotoIf(“PJSIP/anonymous-00000049”, “1?noanonymous”) in new stack
30 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx_builtins.c: Goto (from-sip-external,s,5)
31 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:5] Set(“PJSIP/anonymous-00000049”, “TIMEOUT(absolute)=15”) in new stack
32 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] func_timeout.c: Channel will hangup at 2022-04-12 14:05:36.132 -04.
33 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:6] Set(“PJSIP/anonymous-00000049”, “receveip=pjsip,remote_addr”) in new stack
34 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:7] Log(“PJSIP/anonymous-00000049”, "WARNING,“Rejecting unknown SIP connection from 190.6.132.37:5060"”) in new stack
35 [2022-04-12 14:05:21] WARNING[13024][C-00000043] Ext. s: “Rejecting unknown SIP connection from 190.6.132.37:5060”
36 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:8] Answer(“PJSIP/anonymous-00000049”, “”) in new stack
37 [2022-04-12 14:05:21] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:9] Wait(“PJSIP/anonymous-00000049”, “2”) in new stack
38 [2022-04-12 14:05:23] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:10] Playback(“PJSIP/anonymous-00000049”, “ss-noservice”) in new stack
39 [2022-04-12 14:05:23] VERBOSE[13024][C-00000043] file.c: <PJSIP/anonymous-00000049> Playing ‘ss-noservice.ulaw’ (language ‘es’)
40 [2022-04-12 14:05:32] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:11] PlayTones(“PJSIP/anonymous-00000049”, “congestion”) in new stack
41 [2022-04-12 14:05:32] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:12] Congestion(“PJSIP/anonymous-00000049”, “5”) in new stack
42 [2022-04-12 14:05:33] VERBOSE[13024][C-00000043] pbx.c: Spawn extension (from-sip-external, s, 12) exited non-zero on ‘PJSIP/anonymous-00000049’
43 [2022-04-12 14:05:33] VERBOSE[13024][C-00000043] pbx.c: Executing [[email protected]:1] Hangup(“PJSIP/anonymous-00000049”, “”) in new stack
44 [2022-04-12 14:05:33] VERBOSE[13024][C-00000043] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000049’
45

Assuming a pjsip trunk and IP authentication, set Match (Permit) for the trunk to the list of IP addresses from which Tricom can send calls.

If you are using pjsip but with registration, their domain name may resolve to multiple IP addresses; list all of these in Match (Permit).

If using a chan_sip trunk (you shouldn’t) bound to port 5160, configure the Tricom portal to send calls to port 5160.

Anything else, or no luck, post details of your trunk configuration (mask usernames, passwords, etc.)

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