dSIPRouter w/Official Support for PJSIP has been released to BETA

[ma...@dopensource.com]

Hey Ya’ll,

We have finally released a BETA version of dSIPRouter that has official support for PJSIP for pass-thru authentication.  Here’s the highlights:

- Support for PJSIP 2.9 and PJSIP 2.10, which is the SIP Stack in Asterisk

- PJSIP has a broken PATH implementation, which is the crux to having a proxy in front of Asterisk .  So, we handle keeping track of the routing information versus depending on Asterisk

- We performed testing with Yealink and Polycom Phones

- We have tested FusionPBX and FreePBX pass-thru auth on the same instance of dSIPRouter.  This enables dSIPRouter to proxy request for both simultaneously.

* I would like to thank the paying customers that invested in the development!  This was a very difficult thing to track down and figure out!  

It’s in in BETA on the 0.65 branch and tested only on Debian.  You can install it on Debian using these instructions:

apt-get update -y
apt-get install -y git
cd /opt
git clone https://github.com/dOpensource/dsiprouter.git -b v0.65
cd dsiprouter
./dsiprouter.sh install -all

[Olivier Roy]

I'm testing PJSIP pass-thru on a v0.65 dSIPRouter server. The register gets sent to the FreePBX server that is using PJSIP extensions but it replies with 404 Not Found. If I forward to a FreePBX server using SIP extensions, it works.

[Olivier Roy]

Edit: The problem was because the dSIPRouter instance was configured as a PJSIP trunk on the FreePBX server. This seems to cause issues with the PJSIP endpoint lookup. I deleted the trunk and added the dSIPRouter's IP to the FreePBX firewall. This means that we wouldn't be able to use the same dSIPRouter server for trunking and pass-thru. I tried changing the endpoint identifier order so that it would use the username for lookup but it still didn't work.

[Olivier Roy]

Edit 2: I managed to get path support activated for a PJSIP extension in FreePBX. You have to edit /etc/asterisk/pjsip.aor_custom_post.conf

In my case I added:

[1234](+type=aor)

support_path=true

1234 is my extension number. Afterwards run fwconsole r to reload. If I run asterisk -rx "pjsip aor show 1234", I see that support_path is now set to true.

dSIPRouter sends the path during registration.

Path: <sip:12...@dsip.ip.address;lr;received=sip:zoiper.ip.address:48356>

However if I check the endpoint's contact information in FreePBX, it uses the dSIPRouter IP address as a contact.

[AirsayLongCon]

Was this issue ever resolved? Still experiencing it on v0.72. With a PJSIP dsiprouter trunk, extensions cannot not register to the PBX in Pass Through mode (FreePBX) returnng a 400 Not Found Error. Extensions are able to register when te PJSIP trunk is disabled. Any advice is appreciated

BR

airsay

[Micah Quinn]

I've been able to get pass through mode to work with my pjsip extensions using version 0.71.  I'm using TLS but it registration also seems to work via TCP. Let me know if you want me to provide more details about my configuration.

[AirsayLongCon]

Please go ahead and share

[Micah Quinn]

I installed dSIPRouter on a Debian 11 VM with a single NIC connected to the same LAN as the PBXs. I port forwarded 5060/5061 and 10k-20k through the firewall to the VM. I configured a domain and a pass through endpoint to match the PBX. I can't recall if I had to enable TLS in the kamailio cfg file or not.

I've had issues with the calls disconnecting possibly because of NATing. Disabling WITH_USRLOCDB seems to fix it. I'm still not 100% sure that my setup is stable, but clients to auth and register through to the PBX.

[AirsayLongCon]

So I have my setup mostly similar to yours. I also have a trunk to dsiprouter on FreePBX. When this trunk is enabled, my phones aren't able to register to FreePBX. I get a 404 Error when the phones try to register. When the trunk is disabled, the phones register without issues. I'd like to understand what about by trunk configuration is causing this error. I will share my trunk settings and the sngrep screenshot of the  404 error in my next message

[AirsayLongCon]

Please see screenshots below for SIP trace

[AirsayLongCon]

FreePBX Trunk Settings