I tried to configure my GW1202 as sip trunk in web gui after reading a lot of info, but looks like i'm missing something here...
I set up 2 sip trunks and routing rules for incoming and outgoing calls and what i have is:
Incoming calls from first configured trunk (707) works fine
Incoming calls from second configured trunk (708) gives error in asterisk:
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WARNING[25001][C-0000000d]: chan_sip.c:17169 check_auth: username mismatch, have <708>, digest has <707>
NOTICE[25001][C-0000000d]: chan_sip.c:26213 handle_request_invite: Failed to authenticate device "+XXXXXXXXXXXX" <sip:[email protected]>;tag=as6dab6672
But why?
And when i do outgoing call i have this in asterisk log:
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Executing [707XXXXXXXXXX@call_from_preset_gateway:6] Dial("SIP/212-0000001c", "SIP/707/XXXXXXXXXX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/707/XXXXXXXXXX
-- Got SIP response 503 "Service Unavailable" back from 172.16.11.41:5060
-- SIP/707-0000001d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/212-0000001c' status is 'CONGESTION'
Sorry for the trouble.Can you give me remote access for a check?Teamviewer(version 11) or AnyDesk is ok.
It is very convenient for us to contact via skype.My skype: This email address is being protected from spambots. You need JavaScript enabled to view it.
When you are ready,you can contact me.I will help you to check it.
For others to know, u need to set:
On asterisk sip users configuration:
fromuser=[username]
on Ovox:
voip -> voip endpoints -> edit endpoint ->Advanced:Registration Options -> From User:
set to [username], same as in your asterisk, and enable checkbox 'modify' nearby 'From User:' parameter