It requires a custom plugin.
Let you discuss it with support@brekeke.com.

What does "same parameters" mean?
Same SIP header value? same To-URI and/or same Request line?

It seems that the SIP Server doesn't know whether "neo-vocalcom.be" is the SIP server itself or not. So the SIP Server might try to forward REGISTER to "neo-vocalcom.be". Is the SIP server running behind a NAT? Did you set the SIP Server's public side IP address at [Interface address 1] in the ...

uhupfeld, > NATted calls not working anymore Did the issue start happening after you upgraded the SIP Server? Are you using the same configuration/DialPlan as the before the upgrade? Did you change VPN's routing settings? > Wireshark shows the target packets being sent to the private IP (192.168.15 ...

Glad!

Just use $replaceuri instead of $replaceuri.from and $replaceuri.to.

$replaceuri = false

Do you want to keep the original From/To URIs?
Let you use following DialPlan variables in Deploy Patterns.
$replaceuri.from = false
$replaceuri.to = false

FYI:
https://docs.brekeke.com/sip/replaceuri-from
https://docs.brekeke.com/sip/replaceuri-to

> 111%40flytelecom@172.17.108.122 It is weird. Does REGISTER packet sent from Zoiper indicate such a URL in Contact header? If so, it will be a configuration issue at Zoiper. I'm not sure the following DialPlan rule works or not.. but let you try. Matching Patterns $request = ^INVITE $getSIPuser ...

Do you want to remove a domain name part from the From URI?
Is it SIP-URL? or Tel-URL?

How does the SIP server connect to PSTN? Are you using an FXO gateway? or are you using a telephony provider?

Glad to know it

> The registration in zoiper looks like this: user@domain@mySipIP:5060

Set the "user@domain" at [user / user@host] field.
and set the "mySipIP:5060" at [Domain / Outbound proxy] field.

Have you added a domain at [Domains] page? You will find [New Domain] button there for adding new domain. Also, add new users at [User Authentication] page for the new domain. A value of User filed must be user@domain. For example, if username is 1234 and domain name is test.com, it should be 1234 ...

It depends on a SIP UAC whether it supports DTLS or not.

If you send a call through Brekeke PBX, the PBX can insert DTLS behalf of UAC.

Have you enabled Brekeke PBX to use WebRTC?
https://docs.brekeke.com/pbx/setting-up ... ser-webrtc


Which WebRTC client are you using?

Brekeke uses several information as machine's signature. A MAC address is one of them.

Have you contacted to Brekeke's team? You can ask them why it happened on your license.

FYI:
https://docs.brekeke.com/sip/brekeke-log-level

> to include Status level, would that include the reason phrase in the log?

It will be: net.sip.loglevel.file = 16

You don't have to enable debug logging just for getting responses code and their reasons. Please go to Brekeke PBX's [Diagnostics]->[Debug Logs] page and click [Download] button. After you unzip the downloaded file, you will find "call xxx .log" under pbx/logs/2019/02 The log format is: session-id ...

Which WebRTC based SIP client are you using? Generally WebRTC client can discover a way to exchange RTP packets automatically with ICE so you don't have to use "RTP-Relay". Try the following settings at [Configuration]->[RTP] page. [RTP relay] = "auto" [RTP relay even with ICE] = "no" If it doesn't ...

Currently you need to tune OS's route table to meet your requirement.

How about tuning of the routing table at Linux?

route add -net <UAC's netaddress> netmask 255.255.255.0 dev em2
route add -net <UAS's netaddress> netmask 255.255.255.0 dev em3

If you want to insert users data into the table directly, let you build own Authentication Plug-in.

http://www.brekeke.com/doc/sip/sip_auth ... plugin.pdf

For using TCP, let you upgrade your Brekeke PBX to the latest version because TCP connectivity was improved. http://www.brekeke.com/downloads/pbx.php http://wiki.brekeke.com/wiki/BPBX_Minor_Update Also adjust REGISTER's interval with a shorter number such as 120 (=2min). The setting is [Adjusted ...

> TCP gives upstream problems, after a while no calls come in anymore. It seems there is another issue with TCP... > The UDP packets are from both sides, marked as Don't fragment. For Linux, you may need to tune iptables or use sysctl command for resetting DF flag. You will find some solutions by ...

> FYI, uat-sgvivobeesg.ngtnetworks.com resolved correctly to the Brekeke SIP Proxy server. But the log indicate the following. > register.25: unknown host: sip:uat-sgvivobeesg.ngtnetworks.com:6060 Can you execute the command below at [Tool]]->[Command]? dnslookup uat-sgvivobeesg.ngtnetworks.com

Was the system working previously?

If you are using Internet Explorer as the web-browser, let you refer to this wiki page.
http://wiki.brekeke.com/wiki/IE-doesnt- ... admin-page