[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

Asghar Mohammad asghar144 at gmail.com
Sun Aug 18 12:05:32 CDT 2013


just remove username.
type peer authenticate by ip


On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin <andrew at vsave.co.za> wrote:

>  change server two to host = dynamic
>
> then add register =XXXX on server 1
>
> On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:
>
> Even I tried the type as friend.. but no use...
>
>
> On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N <
> gopalakrishnan.an at gmail.com> wrote:
>
>> Hi,
>>
>>  Am making a simple SIP trunk between two Asterisk server,
>>
>>  Server 1
>> sip.conf
>>  [usman02]
>> type=peer
>> username=usman02
>> secret=usman02
>> host=10.30.2.58
>> context=man02-trunk
>> port=5060
>> qualify=yes
>> disallow=all
>> ;allow=g729
>> allow=g729
>> ;allow=alaw
>> nat=force_rport,comedia
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>> insecure=invite,port
>>
>>  extensions.conf
>>  [man02-trunk]
>> exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
>> exten => _1X.,n,Hangup
>>
>>
>>  Server2
>> sip.conf
>>  [usman02]
>> type=peer
>> username=usman02
>> secret=usman02
>> host=10.10.10.81
>> context=us02-trunk-inbound
>> port=5060
>> qualify=yes
>> disallow=all
>> allow=g729
>> ;allow=ulaw
>> ;allow=alaw
>> nat=force_rport,comedia
>> dtmfmode=rfc2833
>> relaxdtmf=yes
>>  insecure=port,invite
>>
>>  extensions.conf
>>  [us02-trunk-inbound]
>> exten => _X.,Dial(SIP/${EXTEN},60)
>>
>>
>>  Now when I dial from server1, in the server 2 am getting the error as,
>> [Aug 18 09:22:49] WARNING[2779][C-000008db]: chan_sip.c:16266 check_auth:
>> username mismatch, have <2001>, digest has <usman02>
>>
>>  things are fine.. but I dont know where the mistake is...!
>>
>>  Can you some one advise me... !
>>
>>  Thanks.
>>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130818/4db08b56/attachment.htm>


More information about the asterisk-users mailing list