1. Brekeke Product Name and version:Bpbx
2. Java version:2.4.7.3
3. OS type and the version:win 2003 standard
4. UA (phone), gateway or other hardware/software involved:Grandstream GXP280, Asterisk Gateway
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:I've just installed version 2.4.7.3 but can't send calls to an ITSP from the Pbx.
My UA is Grandstream Gxp280 ip phone. UA can register to the server with a Username and password which I also added in the Sip Server "User Authentication".As the UA can not handle "invite Authentication",I added this dial plan in the Sip Server to take care of Proxy Authentication since "request","invite" are enabled.
[Matching Patterns]
$request = ^INVITE
Grandstream_ip
[Deploy Patterns]
$auth=false
$continue=true.
On the Pbx ARS,I have this dial plan.
ARS PATTERN - IN
Nothing,just blank.I assume I don't have to define any rule here as I'm not sending calls to pbx extention.
ARS PATTERN - OUT
[Matching Patterns]
To:sip:(.+)@
[Deploy Patterns]
To:sip:$1@ my_ITSP_IP
With the above dial plan I'm getting 486 busy response.Wireshark trace shows that the call is not reaching the Pbx.It goes to the Sip server and the UA (Grandstream ip phone) returns busy response.
I've also tried to use this dial plan to send calls to the pbx but it didn't work:
[Matching Patterns]
$request=^INVITE
$addr=Grandstream_ip
[Deploy Patterns]
$auth=false
$target=localhost:15060.
I believe there's something I'm missing out.I'd appreciate if someone could clue me in.
Thanks,
Vincent.
2. Java version:2.4.7.3
3. OS type and the version:win 2003 standard
4. UA (phone), gateway or other hardware/software involved:Grandstream GXP280, Asterisk Gateway
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:I've just installed version 2.4.7.3 but can't send calls to an ITSP from the Pbx.
My UA is Grandstream Gxp280 ip phone. UA can register to the server with a Username and password which I also added in the Sip Server "User Authentication".As the UA can not handle "invite Authentication",I added this dial plan in the Sip Server to take care of Proxy Authentication since "request","invite" are enabled.
[Matching Patterns]
$request = ^INVITE
Grandstream_ip
[Deploy Patterns]
$auth=false
$continue=true.
On the Pbx ARS,I have this dial plan.
ARS PATTERN - IN
Nothing,just blank.I assume I don't have to define any rule here as I'm not sending calls to pbx extention.
ARS PATTERN - OUT
[Matching Patterns]
To:sip:(.+)@
[Deploy Patterns]
To:sip:$1@ my_ITSP_IP
With the above dial plan I'm getting 486 busy response.Wireshark trace shows that the call is not reaching the Pbx.It goes to the Sip server and the UA (Grandstream ip phone) returns busy response.
I've also tried to use this dial plan to send calls to the pbx but it didn't work:
[Matching Patterns]
$request=^INVITE
$addr=Grandstream_ip
[Deploy Patterns]
$auth=false
$target=localhost:15060.
I believe there's something I'm missing out.I'd appreciate if someone could clue me in.
Thanks,
Vincent.