1. Brekeke Product Name and version:2.4.7.3/286.1
4. UA (phone), gateway or other hardware/software involved:
Xlite
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : pattern 8
6. Your problem:
I'm using SIP server to send calls to a voip carrier which accepts calls through IP authentication.so traffic is going to voip carrier in following way:
Xlite >> Brekeke SIP server >> voip carrier
also i'm using this dial plan in my sip server:
[matching pattern]
$request=^INVITE
To= sip:00(.+)@
[deploy pattern]
To= sip:%1@CARRIER_IP_ADDRESS
Unfortunately calls rejected from my carrier,Wireshark shows me my carrier sends Forbidden message 403 .
Also I decided to make sure about my carrier to accepts my calls so I tested with my quintum gateway in following way
ISDN >> quintum >> carrier
And it worked for me ,connection established without any error message.I set the quintum register expire time= -1 (means no registration required).
Also I tested my sip server with another voip carrier (voipms).It worked for me too.
But I have problem with my main voip carrier which accept calls though ip authentication.
Any help to solve this problem.
Thanks
Forbidden message 403 from ITSP
Moderator: Brekeke Support Team
B2B-UA Mode turned on but the problem is still there and was not solved.
another thing I want to mention is when i tried to send my calls using Quintum(ISDN>> quintum >> voip carrier) and see my call log, every time calls routed to voip carrier it uses a new ip address of my carrier.It seems carrier reply the sip with new Ip address.
Maybe it can guide you well.
another thing I want to mention is when i tried to send my calls using Quintum(ISDN>> quintum >> voip carrier) and see my call log, every time calls routed to voip carrier it uses a new ip address of my carrier.It seems carrier reply the sip with new Ip address.
Maybe it can guide you well.
Carrier accepts calls using IP authentication so X-lite can't communicate with the my voip carrier because x-lite needs username and password to be registered with server.
But when using quintum to send call to my carier it doesn't have problem.
I guess my carrier can't accept my contact URI because it is different with my "From" section of my dial plan.
My sip server has 2 ip addresses(one public and one private).
Contact URI when calls going out is the private one but "From" contains public IP .
How can i change contact URI to the same as From ip address.
I tried following dial plan but it doesn't works
[Matching pattern]
$request=^INVITE
To=sip:00(.+)@
[Deploy pattern]
To=sip:%1@VOIP_carrier_IP address
From=sip:annonymous@MY_SERVER_PUBLIC_IP
Contact=%1@MY_SERVER_PUBLIC_IP
It is still shown my private ip address in contact URI when capturing by WireShark.
But when using quintum to send call to my carier it doesn't have problem.
I guess my carrier can't accept my contact URI because it is different with my "From" section of my dial plan.
My sip server has 2 ip addresses(one public and one private).
Contact URI when calls going out is the private one but "From" contains public IP .
How can i change contact URI to the same as From ip address.
I tried following dial plan but it doesn't works
[Matching pattern]
$request=^INVITE
To=sip:00(.+)@
[Deploy pattern]
To=sip:%1@VOIP_carrier_IP address
From=sip:annonymous@MY_SERVER_PUBLIC_IP
Contact=%1@MY_SERVER_PUBLIC_IP
It is still shown my private ip address in contact URI when capturing by WireShark.