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Hi chentron
I know Regular expression is not easy to understand but many script languages adapt the syntax of the Regular expression. such as awk, perl and Ruby. Using of DialPlan will be a good cha ...

It seems you are not familiar with Regular expression.
I recommend that you start learning about Regular expression before you edit the DialPlan.
https://en.wikipedia.org/wiki/Regular_expression

...

The above dialplan works with other brand gateways.
Your log looks like, you got a connection timeout.

> gateway_ip:46220

Why your gateway uses 46220 not 5060??

Try this.
[Matching Patte ...

Brekeke SIP Server supports text messaging with SIP MESSAGE method..
Let you try.

chentron, which gateway product are you planning to use?
it depends on a gateway product.

If a gateway supports 1-Stage-Dialing and also makes a register to the SIP server, try this dialplan rule ...

Do you mean "Remote-Party-Id" header?

Try this.
$str.equals($getUri(From), $getUri(Remote-Party-Id)) = false

Do you want to put a dialed number in From header?
Or do you want to keep a caller's number in From header?

The current your dialplan rule puts a dialed number in From header.

> the call comes in, it shows from restricted with no IP address in the "From" Header

See the rule what you defined...

Your "From "definition in Deploy Patterns is the ...

Tom,

> One question I have, will the 404 pop up if the actual handset is turned off?

Which handset product is it?
If a handset is turned off, the Brekeke SIP Server and CISCO Call Manager c ...

Join the Brekeke support or partner program. You can request Brekeke more detailed tech information.
http://www.brekeke.com/support/menu.php
http://www.brekeke.com/partner/

You don't need any DialPlan.

Add the following line in the net.sip.parse.content=true
-----------------

Also, it seems your packet size is larger than 1500, so let you add the following too. ...

I've tested 3CX on Windows 7 with Brekeke SIP Server.
It seems there are no delay issue.

Let you try the latest version of 3CX softphone.

Do you know when the delay happens? Is it in the making a call? Or is it in the receiving a call?

It seems delay is a common issue at 3CX phone...
https://www.google.com/search?q=3cx+delay+site%3 ...

A browser such as IE will not cause any issue in SIP.

Also you can set trusted IP addresses in the router.
If you are using unix-like OS (e.g. Linux), tune the iptables to define trusted IP addresses.

https://isc.sans.edu/port.html?port=5060
There ...

Yes. the payload type 18 means G729.
(See http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml#rtp-parameters-1)

> Where can i see if its blocked?

If the above rule matches, i ...

Are you sure that the INVITE's SDP indicates GSM as a codec?
If not, the above rule doesn't match and "404" will be returned.

Can you paste the SDP part of INVITE here?

Can you paste your rules here?

If you are using Brekeke ver3, set "dialplan.debug.log=true" in the Config's Advanced page for debugging DialPlan.

> $body=( "(m=audio .+ RTP/AVP.* 3 )" ) = .+

It is wrong..
It must be
$body( "(m=audio .+ RTP/AVP.* 3 )" ) = .+

Refer to the following page. Is is the list of codec payload types.
http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml#rtp-parameters-1

Use $body in the Matching Pattern.

$body( "(m=audio .+ RTP/AVP.* 3 )" ) = .+

If a SDP indicates GSM (3 is GMS's payloadtype), the above $body matches.

Sunny,

Disable Authentication at SIP Server's [Configuration] > [SIP] page.
REGISTER = off
INVITE = off

You will the info if you read the quick start guide...

hi RE,

if two network cards are activated, you said one of them is a global IP address.

How about another? Is it a local IP address?
If yes, let me know the IP address.

Do you want to load balance between two Brekeke SIP Servers?