Author |
Message |
juniper Brekeke Member
Joined: 09 Feb 2010 Posts: 13
|
Posted: Wed Mar 24, 2010 11:25 am Post subject: SIP Trunk Incoming cals |
|
|
1. Brekeke Product Name and version: Brekeke SIP Server 2.4.3.9/286
2. Java version:1.5.0_09
3. OS type and the version: Windows 2003 R2 SP2 32 bit 5.2
4. UA (phone), gateway or other hardware/software involved: x-lite (direct SIP trunk form provider)
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :
6. Your problem:
we have got the below DID range from provider 18801495-18801498 (+3618801495....)
the outgoing calls are working fine but we have problem with incoming calls
we are using the below dial plan roles :
PMatching Patterns
$addr=91.83.81.243
$request=^INVITE
$getUri( To )=sip:\18801(.+)
Deploy Patterns | $target = localhost:15060 $auth = false $continue = true To = sip:%1
|
We have a user in brekeke PBX with name 495
we would get the calls on this extension when the 18801495 are called.
we have the following entry in the logs
sip:36709676434@91.83.82.243 sip:18801495@10.222.0.98:5060 00:00:00 Wed Mar 24 19:17:05 CET 2010 Wed Mar 24 19:17:05 CET 2010 Busy 486
if we throw all the calls directly to the extension 495 with the below rules the call gets to the x-lite client where the user 495 is logged on:
PMatching Patterns
$addr=91.83.81.243
$request=^INVITE
$getUri( To )=sip:(.+)
Deploy Patterns | $target = localhost:15060 $auth = false $continue = true To = sip:495
|
but in this case I can use only one extension.
How can I get wok all the dids with multiple users, extensions
Regards
Zoltán |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Thu Mar 25, 2010 2:15 pm Post subject: |
|
|
change your dial plan as below and click apply rule after change
Matching Patterns | $addr = 91.83.81.243 $request = ^INVITE
| Deploy Patterns | $target = localhost:15060 $auth = false $transport = udp $b2bua = true
|
And create ARS pattern-in as
Matching Patterns
To: sip:18801(.+)@
Deploy Patterns
To: $1
If it doesnot work, capture packets of the call on your pbx server pc and check if the source ip of the INVITE is 91.83.82.243 and if To header is like sip:18801...@....
If not same, change dial plan $addr and ars To as what is shown in the packets |
|
Back to top |
|
|