A sip trunk between two asterisk with Asterisk Realtime

Asterisk Asterisk Support
1

Hi,
I can’t make a call through a sip trunk between two asterisk with Asterisk Realtime.Console message is

NOTICE[15984]: chan_sip.c:20605 handle_response_invite: Failed to authenticate on INVITE to '"58034000" <sip:58034000@192.168.11.21:65060>;tag=as01fb8660' -- SIP/DQ-00000001 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)]

I need you help,

thanks

2

Authentication error.
As my “Tarot-mode” is off for now You need to post more information about Your configuration (especially the config of the accounts of the two boxes building the trunk connection). Otherwise more sspecific assistance is impossible.

3

Hi,
here is my configuration information:
SERVER[color=#BF0000] A[/color]
[DQ]
type=peer
port=65060
dtmfmode=rfc2833
disallow=all
allow=alaw,ulaw
host=192.168.11.20
context=did
[a2b]
exten => _580340.,1,Dial(SIP/DQ/${EXTEN},20,r)
exten => _580340.,n,Hangup

SERVER[color=#BF0000] B[/color]
[ZJ]
type=peer
port=65060
dtmfmode=rfc2833
disallow=all
allow=alaw,ulaw
host=192.168.11.21
context=did
[a2b]
exten => _580341.,1,Dial(SIP/ZJ/${EXTEN},20,r)
exten => _580341.,n,Hangup

One X-lite 58034000 Registered to server A ,one X-lite 58034111Registered to server B.
when 58034000 call 58034111, The Console information is :

== Using SIP RTP CoS mark 5
– Executing [58034111@a2billing:1] Dial(“SIP/58034000-00000000”, “SIP/DQ/58034111,20,r”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/DQ/58034111
[Jul 26 09:11:35] NOTICE[15984]: chan_sip.c:20605 handle_response_invite: Failed to authenticate on INVITE to ‘“58034000” sip:58034000@192.168.11.21:65060;tag=as01fb8660’
– SIP/DQ-00000001 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [58034111@a2billing:2] Hangup(“SIP/58034000-00000000”, “”) in new stack

4

Try to include the line

in both peer configs (DQ and ZJ). This should clear the authentication error.

And: What will context did do? Hopefully it distributes the incoming calls :smile:

BTW: The other (more performant) method to have a trunk between two or mor Asterisk-boxes is based on the IAX2-protocol, You should e.g. have a look here: Dual Serveers with IAX2

5

Where have you set the server’s port number?