Dial Plan assistance / understanding
Moderator: Brekeke Support Team
Dial Plan assistance / understanding
1. Brekeke Product Name and Version:
Brekeke SIP Server 2.4.8.6/286.3
2. Java version:
1.6.0_45
3. OS type and the version:
Windows Server 2008 R2
4. UA (phone), gateway or other hardware/software involved:
5. Your problem:
understanding dial plan
Below are 2 dial plans, can anyone tell me what is the difference in them ? i see visually a different, but i dont understand the logic behind it
the first one didnt work for me, so i was hacking around and using the 2nd one now works
anyone able to help me, i would appriciate it alot
Matching Patterns:
$request=^INVITE
From=sip:.+@sip1.name.com
Remote-Party-ID=(.+)@
To=sip:(.+)@
$getUri(From)=@(.+)
$addr=10.0.0.156
Deploy Patterns:
$auth=false
From=%1@%3
To=sip:%2@192.168.1.110
$session=failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting=5
&failover.timer.provisional=360
$continue=false
and this is the 2nd version of this dial plan
Matching Patterns:
$request=^INVITE
From=sip:(.+)@sip1.name.com
To=sip:(.+)@
$getUri(From)=@(.+)
$addr=10.0.0.156|11.0.0.156
Deploy Patterns:
$auth=false
From=%1@%3
P-Asserted-Identity=%1@%3
To=sip:%2@192.168.1.110
$session=failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting=5
&failover.timer.provisional=360
$continue=false
Brekeke SIP Server 2.4.8.6/286.3
2. Java version:
1.6.0_45
3. OS type and the version:
Windows Server 2008 R2
4. UA (phone), gateway or other hardware/software involved:
5. Your problem:
understanding dial plan
Below are 2 dial plans, can anyone tell me what is the difference in them ? i see visually a different, but i dont understand the logic behind it
the first one didnt work for me, so i was hacking around and using the 2nd one now works
anyone able to help me, i would appriciate it alot
Matching Patterns:
$request=^INVITE
From=sip:.+@sip1.name.com
Remote-Party-ID=(.+)@
To=sip:(.+)@
$getUri(From)=@(.+)
$addr=10.0.0.156
Deploy Patterns:
$auth=false
From=%1@%3
To=sip:%2@192.168.1.110
$session=failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting=5
&failover.timer.provisional=360
$continue=false
and this is the 2nd version of this dial plan
Matching Patterns:
$request=^INVITE
From=sip:(.+)@sip1.name.com
To=sip:(.+)@
$getUri(From)=@(.+)
$addr=10.0.0.156|11.0.0.156
Deploy Patterns:
$auth=false
From=%1@%3
P-Asserted-Identity=%1@%3
To=sip:%2@192.168.1.110
$session=failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting=5
&failover.timer.provisional=360
$continue=false
The first one has the following definition in the Matching Patterns.
Remote-Party-ID = (.+)@
It means that "Remote-Party-ID" header is required in INVITE to hit the DialPlan.
Depends on SIP client or SIP trunk, INVITE will not have "Remote-Party-ID" header.
The second one doesn't require "Remote-Party-ID" header.
So.. it will match any kind of INVITE packets.
Remote-Party-ID = (.+)@
It means that "Remote-Party-ID" header is required in INVITE to hit the DialPlan.
Depends on SIP client or SIP trunk, INVITE will not have "Remote-Party-ID" header.
The second one doesn't require "Remote-Party-ID" header.
So.. it will match any kind of INVITE packets.
The second one can be simplified as the following.
[Matching Patterns]
$request = ^INVITE
$getUri(From) = sip:(.+@sip1.name.com)
To = sip:(.+)@
$addr = ^10.0.0.156$|^11.0.0.156$
[Deploy Patterns]
$auth = false
From = sip:%1
P-Asserted-Identity = sip:%1
To = sip:%2@192.168.1.110
$session = failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting = 5
&failover.timer.provisional = 360
$continue = false
You need ^ in the front of IP address at $addr.
$addr = 10.0.0.156|11.0.0.156 will match 110.0.0.156, 111.0.0.156, 210.0.0.156 and 211.0.0.156...
[Matching Patterns]
$request = ^INVITE
$getUri(From) = sip:(.+@sip1.name.com)
To = sip:(.+)@
$addr = ^10.0.0.156$|^11.0.0.156$
[Deploy Patterns]
$auth = false
From = sip:%1
P-Asserted-Identity = sip:%1
To = sip:%2@192.168.1.110
$session = failover sip:%2@192.168.1.110 sip:%2@192.168.1.111
&failover.timer.inviting = 5
&failover.timer.provisional = 360
$continue = false
You need ^ in the front of IP address at $addr.
$addr = 10.0.0.156|11.0.0.156 will match 110.0.0.156, 111.0.0.156, 210.0.0.156 and 211.0.0.156...
I need to follow this instructions, and i need help how to implement it
https://support.vampcommunications.com/ ... nformation
also this one
https://support.vampcommunications.com/ ... E-examples
if anyone is able to help me how to setup my dial plan, i would really appreciate it
https://support.vampcommunications.com/ ... nformation
also this one
https://support.vampcommunications.com/ ... E-examples
if anyone is able to help me how to setup my dial plan, i would really appreciate it
Caller ID / Privacy Header Information
Q. When sending calls through SIPRotues, is it necessary for our system to include the P-Asserted-Identity or Remote-Party-ID fields in the header of the INVITE?
A. These fields are not required. If you do not pass one of these, we will automatically populate the appropriate header with your FROM header information depending on what the upstream carrier wants. However if you wish to control privacy you would use either Remote-Party-ID or P-Asserted-Identity. We prefer you to send P-Asserted-Identity. See below for formatting.
Q: How can I block caller id?
A: To block caller ID you would add one of the following headers (change to match your originating ANI). The privacy header tells the far end to mask caller ID.
(E.164 format or 10 digits required, do NOT use 11 digits)(we prefer P-Asserted-Identity and so do our carriers) (PAI takes presidence over Remote-Party-ID)
P-Asserted-Identity: "Sheldon Cooper" <sip:+12125551212@4.4.4.4>
Privacy: id
or
Remote-Party-ID: “Sheldon Cooper” <sip:+19195551212@4.4.4.4>;party=calling;screen=yes;privacy=full
You can NOT relay privacy by only sending a From header.
From: "Sheldon Cooper" <sip:+12125551212@4.4.4.4>
If you would like to review the RFC on this please visit this link:
http://www.ietf.org/rfc/rfc3325.txt
Additional INFO:
Some carriers WILL pass along the "UNKNOWN" field in a FROM/Remote-Party-Id/P-Asserted-Id header. Therefore it is recommended that you NOT put UNKNOWN in any of these headers. Standard procedure is for the terminating carrier to dip the ANI and provide the correct name, however, some carrier switches view the "UNKNOWN" as a privacy attempt and pass it along or do not terminate the call.
Example of what NOT to do:
From: "Unknown" sip:+19198900000@72.15.219.140;user=phone
Remote-Party-Id: "Unknown" sip:+19198900000@72.15.219.140;user=phone
P-Asserted-Identity: "Unknown" sip:+19198900000@72.15.219.140;user=phone
Example of what TO
From: "name" sip:+19198900000@72.15.219.140;user=phone
Remote-Party-Id: "name" sip:+19198900000@72.15.219.140;user=phone
P-Asserted-Identity: "name" sip:+19198900000@72.15.219.140;user=phone
INVITE examples
Example 2: sending 11 digits, E.164, no prefix
INVITE sip:+19198900000@72.15.219.140 SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK22f387ce;rport.
From: "SIPRoutes" <sip:9198900000@xxx.xxx.xxx.xxx>;tag=as08310909.
To: <sip:19198900000@72.15.219.140>.
Contact: <sip:9198900000@xxx.xxx.xxx.xxx>.
Call-ID: 1ea01a2f16e135cd1e6cacaf5e79202a@xxx.xxx.xxx.xxx.
CSeq: 102 INVITE.
User-Agent: Vamp UAC.
Max-Forwards: 70.
Date: Wed, 25 Apr 2012 16:14:16 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 3226 3226 IN IP4 xxx.xxx.xxx.xxx.
s=session.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 11444 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
Q. When sending calls through SIPRotues, is it necessary for our system to include the P-Asserted-Identity or Remote-Party-ID fields in the header of the INVITE?
A. These fields are not required. If you do not pass one of these, we will automatically populate the appropriate header with your FROM header information depending on what the upstream carrier wants. However if you wish to control privacy you would use either Remote-Party-ID or P-Asserted-Identity. We prefer you to send P-Asserted-Identity. See below for formatting.
Q: How can I block caller id?
A: To block caller ID you would add one of the following headers (change to match your originating ANI). The privacy header tells the far end to mask caller ID.
(E.164 format or 10 digits required, do NOT use 11 digits)(we prefer P-Asserted-Identity and so do our carriers) (PAI takes presidence over Remote-Party-ID)
P-Asserted-Identity: "Sheldon Cooper" <sip:+12125551212@4.4.4.4>
Privacy: id
or
Remote-Party-ID: “Sheldon Cooper” <sip:+19195551212@4.4.4.4>;party=calling;screen=yes;privacy=full
You can NOT relay privacy by only sending a From header.
From: "Sheldon Cooper" <sip:+12125551212@4.4.4.4>
If you would like to review the RFC on this please visit this link:
http://www.ietf.org/rfc/rfc3325.txt
Additional INFO:
Some carriers WILL pass along the "UNKNOWN" field in a FROM/Remote-Party-Id/P-Asserted-Id header. Therefore it is recommended that you NOT put UNKNOWN in any of these headers. Standard procedure is for the terminating carrier to dip the ANI and provide the correct name, however, some carrier switches view the "UNKNOWN" as a privacy attempt and pass it along or do not terminate the call.
Example of what NOT to do:
From: "Unknown" sip:+19198900000@72.15.219.140;user=phone
Remote-Party-Id: "Unknown" sip:+19198900000@72.15.219.140;user=phone
P-Asserted-Identity: "Unknown" sip:+19198900000@72.15.219.140;user=phone
Example of what TO
From: "name" sip:+19198900000@72.15.219.140;user=phone
Remote-Party-Id: "name" sip:+19198900000@72.15.219.140;user=phone
P-Asserted-Identity: "name" sip:+19198900000@72.15.219.140;user=phone
INVITE examples
Example 2: sending 11 digits, E.164, no prefix
INVITE sip:+19198900000@72.15.219.140 SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK22f387ce;rport.
From: "SIPRoutes" <sip:9198900000@xxx.xxx.xxx.xxx>;tag=as08310909.
To: <sip:19198900000@72.15.219.140>.
Contact: <sip:9198900000@xxx.xxx.xxx.xxx>.
Call-ID: 1ea01a2f16e135cd1e6cacaf5e79202a@xxx.xxx.xxx.xxx.
CSeq: 102 INVITE.
User-Agent: Vamp UAC.
Max-Forwards: 70.
Date: Wed, 25 Apr 2012 16:14:16 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 3226 3226 IN IP4 xxx.xxx.xxx.xxx.
s=session.
c=IN IP4 xxx.xxx.xxx.xxx.
t=0 0.
m=audio 11444 RTP/AVP 0 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
so... what do you want?
> Example of what NOT to do:
> From: "Unknown" sip:+19198900000@72.15.219.140;user=phone
> Example of what TO
> From: "name" sip:+19198900000@72.15.219.140;user=phone
Try the following definition in the Deploy Pattern.
From/displayname = "name"
It replaces the From-header's display name with "name".
> Example of what NOT to do:
> From: "Unknown" sip:+19198900000@72.15.219.140;user=phone
> Example of what TO
> From: "name" sip:+19198900000@72.15.219.140;user=phone
Try the following definition in the Deploy Pattern.
From/displayname = "name"
It replaces the From-header's display name with "name".