Asterisk returns everyone is busy/congested yet all my gateway ports are open

Asterisk Asterisk SIP
1

[general]
context=from-sip
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=g729

; ===========================================
; ORIGINATOR (fixed IP)
; ===========================================
[originator]
type=friend
host=88.171.77.276
context=from-internal
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
qualify=yes
nat=no
permit=88.171.77.276/255.255.255.255
deny=0.0.0.0/0.0.0.0

; ===========================================
; GATEWAY (GoIP8) - Dynamic IP, registers to Asterisk
; ===========================================
[Goip-8]
type=friend
host=dynamic
context=from-sip
username=goipuser
secret=goippass
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
qualify=yes
nat=yes

[from-internal]

; ===========================================
; Incoming calls from ORIGINATOR with tech prefix 888
; ===========================================
exten => _8882567XXXXXXXX,1,NoOp(Call flow: fake_ringback + speech detect + prefix strip)
same => n,Answer()

; Strip 888 prefix
same => n,Set(DESTNUM=${EXTEN:3})
same => n,NoOp(Stripped number is ${DESTNUM})

; Step 1: Detect speech during fake_ringback playback
same => n,BackgroundDetect(fake_ringback,,100,,talk)

; Step 2: Detect speech during 10-second silent audio
same => n,BackgroundDetect(silence10s,,100,,talk)

; Step 3: If no speech detected after fake_ringback + silence, hang up
same => n,NoOp(No speech detected after fake_ringback + 10s)
same => n,Hangup(34)

; ===========================================
; Speech detected - forward to Goip-8 gateway
; ===========================================
exten => talk,1,NoOp(Speech detected! Forwarding to Goip-8 gateway)
same => n,Dial(SIP/${DESTNUM}@Goip-8,30,m(myringback))
same => n,Hangup()

== Using SIP RTP CoS mark 5
– Executing [888256701076492@from-internal:1] NoOp(“SIP/test-00000018”, “Ca ll flow: fake_ringback + speech detect + prefix strip - ANY 888 number”) in new stack
– Executing [888256701076492@from-internal:2] Answer(“SIP/test-00000018”, " ") in new stack
– Executing [888256701076492@from-internal:3] Set(“SIP/test-00000018”, “DES TNUM=256701076492”) in new stack
– Executing [888256701076492@from-internal:4] NoOp(“SIP/test-00000018”, “St ripped number is 256701076492”) in new stack
– Executing [888256701076492@from-internal:5] BackgroundDetect(“SIP/test-00 000018”, “fake_ringback,100,talk”) in new stack
– <SIP/test-00000018> Playing ‘fake_ringback.ulaw’ (language ‘en’)
– Executing [888256701076492@from-internal:6] BackgroundDetect(“SIP/test-00 000018”, “silence10s,100,talk”) in new stack
– <SIP/test-00000018> Playing ‘silence10s.ulaw’ (language ‘en’)
– Executing [talk@from-internal:1] NoOp(“SIP/test-00000018”, “Speech detect ed! Forwarding to Goip-8 gateway”) in new stack
– Executing [talk@from-internal:2] Dial(“SIP/test-00000018”, “SIP/256701076 492@Goip-8,30,m(myringback)”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/256701076492@Goip-8
– Started music on hold, class ‘myringback’, on channel ‘SIP/test-00000018’
== Everyone is busy/congested at this time (1:0/0/1)
– Stopped music on hold on SIP/test-00000018
– Executing [talk@from-internal:3] Hangup(“SIP/test-00000018”, “”) in new s tack
== Spawn extension (from-internal, talk, 3) exited non-zero on ‘SIP/test-00000 018’

2

You’d need to actually show SIP traffic from “sip set debug on” to see what the remote side has returned, if an attempt was made.

3

You have specified that there are no acceptable source addresses! Also you are using an obsolete channel driver which has not been included in recent major versions of the Asterisk source code. I think this only affects incoming calls, so probably isn’t the cause of the current problem.

4

Okay so i should use pjsip

And what’s the address field supposed to look like?

5

I’m not sure which address field you are referring to, but assuming it relates to deny, permit and deny are processed in order of appearance, so your deny cancels out the permit.

Also, with chan_sip you using type=peer, which is almost certainly better here, means that only the host= line, and not the user name, will be used to identify callers, so you wouldn’t need permit and deny as well, as you would already be limited to only one IP address. In both cases, your insecure=port (which is normally there because someone copied and pasted, code that had been blindly changed from insecure=very) means that you are not restricting the source port number, even though providers almost always only use the standard one.

6

Okay thanks

Let me remove the line “deny=0.0.0.0/“

And change insecure=very

Then try again

7

insecure=very will be ignored. It was removed to discourage people from blindly specifying it, but they simply translated it to the equivalent full set of options, rather than thinking about what they really needed to make insecure.

8

A SIP trace as previously mentioned is really needed.

9

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