Unable to make TLS call

Asterisk Asterisk Support
1

After calling a number, it’s connecting always, then give me a Service Unavailable 1 minute later. The Asterisk 1.8 works well when using UDP. Could anybody help me? AST.pdf does not help me…

== Using SIP RTP CoS mark 5
– Executing [1098@phones:1] Set(“SIP/1098-00000079”, “_SIPSRTP_CRYPTO=enable”) in new stack
– Executing [1098@phones:2] Dial(“SIP/1098-00000079”, “SIP/1098”) in new stack
== Using SIP RTP CoS mark 5
– Called 1098
– SIP/1098-0000007a is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/1098-00000079’ status is 'CONGESTION’
asterisk18*CLI>

exten => 1098,1,Set(_SIPSRTP_CRYPTO=enable)
exten => 1098,2,Dial(SIP/${EXTEN})
exten => 1099,1,Set(_SIPSRTP_CRYPTO=enable)
exten => 1099,2,Dial(SIP/${EXTEN})
in extensions.conf

[1098]
type=friend
username=1098
secret=1
host=dynamic
context=phones
callerid=“secure” <1098>
encryption=yes
transport=tls
port=5061
[1099]
type=friend
username=1099
secret=1
host=dynamic
context=phones
callerid=“secure” <1099>
encryption=yes
transport=tls
port=5061
in sip.conf

2

wiki.asterisk.org/wiki/display/ … g+Tutorial

and

wiki.asterisk.org/wiki/display/ … +Specifics

Those functions you’re trying to call don’t exist; that’s presumably from the SRTP patch before it was merged.

Cheers.

3

Still failed, what else should be done?

*CLI> == Using SIP RTP CoS mark 5
[Mar 15 16:18:36] WARNING[6995]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): s
yntax error: syntax error, unexpected ‘’, expecting $end; Input:
1" = “”
^
[Mar 15 16:18:36] WARNING[6995]: ast_expr2.fl:472 ast_yyerror: If you have quest
ions, please refer to wiki.asterisk.org/wiki/display/ … el+Variabl
es
– Executing [1098@phones:1] GotoIf(“SIP/1098-00000001”, "“1?fail”) in new s
tack
– Goto (phones,1098,5)
– Executing [1098@phones:5] Playback(“SIP/1098-00000001”, “vm-goodbye”) in
new stack
– <SIP/1098-00000001> Playing ‘vm-goodbye.gsm’ (language ‘en’)
– Executing [1098@phones:6] Hangup(“SIP/1098-00000001”, “”) in new stack
== Spawn extension (phones, 1098, 6) exited non-zero on ‘SIP/1098-00000001’

[1098]
type=peer
secret=1
host=dynamic
context=phones
dtmfmode=rfc2833
disallow=all
allow=g722
transport=tls
encryption=yes

in sip.conf

exten => 1098,1,GotoIf("$[${CHANNEL(secure_signaling)}" = “”]?fail)
exten => 1098,n,GotoIf("$[${CHANNEL(secure_media)}" = “”]?fail)
exten => 1098,n,Dial(SIP/1098)
exten => 1098,n,Hangup
exten => 1098,n(fail),Playback(vm-goodbye)
exten => 1098,n,Hangup

in extension.conf

4

I have modified the example you are using from:

wiki.asterisk.org/wiki/display/ … +Specifics

Please update and try again.

Cheers.

5

failed again,

*CLI> [Mar 16 09:53:03] NOTICE[9158]: chan_sip.c:23453 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1098
== Using SIP RTP CoS mark 5
[Mar 16 09:53:03] WARNING[9163]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘’, expecting $end; Input:
1" = “1”
^
[Mar 16 09:53:03] WARNING[9163]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to wiki.asterisk.org/wiki/display/ … +Variables
– Executing [1098@phones:1] GotoIf(“SIP/1098-00000003”, "“1?fail”) in new stack
– Goto (phones,1098,5)
– Executing [1098@phones:5] Playback(“SIP/1098-00000003”, “vm-goodbye”) in new stack
– <SIP/1098-00000003> Playing ‘vm-goodbye.gsm’ (language ‘en’)
– Executing [1098@phones:6] Hangup(“SIP/1098-00000003”, “”) in new stack
== Spawn extension (phones, 1098, 6) exited non-zero on ‘SIP/1098-00000003’

6

I don’t think you updated per my updated example. I see this line:

Executing [1098@phones:1] GotoIf("SIP/1098-00000003", ""1?fail") in new stack

When you should be seeing something like:

Executing [123@testing:1] GotoIf("SIP/malcolm-00000000", "0?:fail") in new stack

In my case, I’m testing for secure signaling from a client that’s configured to use udp and it’s failing.

Cheers.

7

failed again…

CLI>
*CLI> [Mar 17 20:05:07] NOTICE[13249]: chan_sip.c:23453 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1098
== Using SIP RTP CoS mark 5
[Mar 17 20:05:09] WARNING[13297]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘’, expecting $end; Input:
1" = “1”
^
[Mar 17 20:05:09] WARNING[13297]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to wiki.asterisk.org/wiki/display/ … +Variables
– Executing [1098@phones:1] GotoIf(“SIP/1098-00000014”, ““1?:fail”) in new stack
[Mar 17 20:05:09] WARNING[13297]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘’, expecting $end; Input:
1” = “1”
^
[Mar 17 20:05:09] WARNING[13297]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to wiki.asterisk.org/wiki/display/ … +Variables
– Executing [1098@phones:2] GotoIf(“SIP/1098-00000014”, "“1?:fail”) in new stack
– Executing [1098@phones:3] Dial(“SIP/1098-00000014”, “SIP/1098”) in new stack
== Using SIP RTP CoS mark 5
– Called 1098
– SIP/1098-00000015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [1098@phones:4] Hangup(“SIP/1098-00000014”, “”) in new stack
== Spawn extension (phones, 1098, 4) exited non-zero on ‘SIP/1098-00000014’
;5~[Mar 17 20:06:04] ERROR[13295]: tcptls.c:375 ast_tcptls_client_start: Unable to connect SIP socket to 9.123.145.239:2399: Connection timed out

8

Hi,

;5~[Mar 17 20:06:04] ERROR[13295]: tcptls.c:375 ast_tcptls_client_start: Unable to connect SIP socket to 9.123.145.239:2399: Connection timed out

Do you actually have TLS enabled in sip.conf? Do you have certificates generated and installed?

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