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Questions about VS-GW2120/GW1600/GW1202 and WGW1002G GSM Series VoIP Gateway
sip trunk configuration issues
7 years 2 weeks ago #11489
by Nufay
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:
But why?
And when i do outgoing call i have this in asterisk log:
I can't understand how to configure it right.
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:
Warning: Spoiler!
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[ Click to hide ]
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=as6dab6672But why?
And when i do outgoing call i have this in asterisk log:
Warning: Spoiler!
[ Click to expand ]
[ Click to hide ]
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'I can't understand how to configure it right.
7 years 2 weeks ago #11490
by heizi.lv
Hi
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.
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.
7 years 2 weeks ago #11495
by Nufay
For others to know, u need to set:
On asterisk sip users configuration: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
Regards,
Nufay.
On asterisk sip users configuration:
fromuser=[username]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
Regards,
Nufay.
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