[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers
Gopalakrishnan N
gopalakrishnan.an at gmail.com
Sun Aug 18 13:32:13 CDT 2013
ok thanks Asghar Mohammad
On Mon, Aug 19, 2013 at 1:05 AM, Asghar Mohammad <asghar144 at gmail.com>wrote:
> 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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/20130819/813f0dea/attachment.htm>
More information about the asterisk-users
mailing list