All,
I've tried searching and have found similar posts before, but I have yet to find a real 'solution'.
I'm running Kamailio 6.0.1 & RTPEngine 13.3.1.7 (on a Linux VM) as a "SBC" (at home) using the config from: https://github.com/voiceboys/sbcOS/blob/version_2.0/var/www/html/alpine/conf...
Except that I've added the following:
mhomed=1 modparam("rr", "force_send_socket", 1)
The SBC is dual-homed: one interface on the local LAN and another interface directly on the Internet/WAN (real public IP).
Everything works fine, EXCEPT if the remote end (the side on the WAN/Internet) hangs up first. If they do, the BYE never makes it past Kamailio to the local PBX and/or phone.
https://users.openser.narkive.com/1aK7n8Il/sr-multihomed-kamailio-and-enable... to use $sht to save the information needed, but I have no clue where in the config to put that.
Any help would be greatly appreciated.
Thanks!
-Dustin
Will be fine to check SIP messages. Here is important Route, Record-Route and contact header values.
Sergey.
On Mon, 2025-06-23 at 20:24 -0500, Dustin Marquess via sr-users wrote:
All,
I've tried searching and have found similar posts before, but I have yet to find a real 'solution'.
I'm running Kamailio 6.0.1 & RTPEngine 13.3.1.7 (on a Linux VM) as a "SBC" (at home) using the config from: https://github.com/voiceboys/sbcOS/blob/version_2.0/var/www/html/alpine/conf...
Except that I've added the following:
mhomed=1 modparam("rr", "force_send_socket", 1)
The SBC is dual-homed: one interface on the local LAN and another interface directly on the Internet/WAN (real public IP).
Everything works fine, EXCEPT if the remote end (the side on the WAN/Internet) hangs up first. If they do, the BYE never makes it past Kamailio to the local PBX and/or phone.
https://users.openser.narkive.com/1aK7n8Il/sr-multihomed-kamailio-and-enable... suggests to use $sht to save the information needed, but I have no clue where in the config to put that.
Any help would be greatly appreciated.
Thanks!
-Dustin __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
Just to add here, double rr with putting the interface details in them using the record_route_preset ( https://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.f.record_rout...) might be the way to go here.
This will allow the BYE to have the correct routes to get back out hopefully.
So on initial invite from private: in rr header 1st rr is from public, 2nd rr is private and on initial invite from public: in rr header 1st rr is from private, 2nd rr is public
This orders the rr headers so on the way back they come in correctly if that makes sense.
I do this at the initial Invite and it works fine for me, but your mileage may vary.
Hope this helps,
John.
On Tue, 24 Jun 2025 at 10:25, Sergei Safarov via sr-users < sr-users@lists.kamailio.org> wrote:
Will be fine to check SIP messages. Here is important Route, Record-Route and contact header values.
Sergey.
On Mon, 2025-06-23 at 20:24 -0500, Dustin Marquess via sr-users wrote:
All,
I've tried searching and have found similar posts before, but I have yet to find a real 'solution'.
I'm running Kamailio 6.0.1 & RTPEngine 13.3.1.7 (on a Linux VM) as a "SBC" (at home) using the config from:
https://github.com/voiceboys/sbcOS/blob/version_2.0/var/www/html/alpine/conf...
Except that I've added the following:
mhomed=1 modparam("rr", "force_send_socket", 1)
The SBC is dual-homed: one interface on the local LAN and another interface directly on the Internet/WAN (real public IP).
Everything works fine, EXCEPT if the remote end (the side on the WAN/Internet) hangs up first. If they do, the BYE never makes it past Kamailio to the local PBX and/or phone.
https://users.openser.narkive.com/1aK7n8Il/sr-multihomed-kamailio-and-enable... suggests to use $sht to save the information needed, but I have no clue where in the config to put that.
Any help would be greatly appreciated.
Thanks!
-Dustin __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users@lists.kamailio.org To unsubscribe send an email to sr-users-leave@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender!
John,
This reply probably isn't going to be helpful, since I obviously misread your entire email. For some reason I was thinking enable_double_rr, so I set:
modparam("rr", "enable_double_rr", 2)
And tried again, same issue.
Sergey asked for the SIP messages. I didn't want to spam the list, so I put them at https://fdf.net/calls.txt
It's late now, I'll try record_route_preset in the morning and try again. 🤦
Thanks! -Dustin
On Tue, Jun 24, 2025 at 5:17 AM Who AmI myfriendjohn1@gmail.com wrote:
Just to add here, double rr with putting the interface details in them using the record_route_preset ( https://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.f.record_rout...) might be the way to go here.
This will allow the BYE to have the correct routes to get back out hopefully.
So on initial invite from private: in rr header 1st rr is from public, 2nd rr is private and on initial invite from public: in rr header 1st rr is from private, 2nd rr is public
This orders the rr headers so on the way back they come in correctly if that makes sense.
I do this at the initial Invite and it works fine for me, but your mileage may vary.
Hope this helps,
John.
On Tue, 24 Jun 2025 at 10:25, Sergei Safarov via sr-users < sr-users@lists.kamailio.org> wrote:
Will be fine to check SIP messages. Here is important Route, Record-Route and contact header values.
Sergey.
On Mon, 2025-06-23 at 20:24 -0500, Dustin Marquess via sr-users wrote:
All,
I've tried searching and have found similar posts before, but I have yet to find a real 'solution'.
I'm running Kamailio 6.0.1 & RTPEngine 13.3.1.7 (on a Linux VM) as a "SBC" (at home) using the config from:
https://github.com/voiceboys/sbcOS/blob/version_2.0/var/www/html/alpine/conf...
Except that I've added the following:
mhomed=1 modparam("rr", "force_send_socket", 1)
The SBC is dual-homed: one interface on the local LAN and another interface directly on the Internet/WAN (real public IP).
Everything works fine, EXCEPT if the remote end (the side on the WAN/Internet) hangs up first. If they do, the BYE never makes it past Kamailio to the local PBX and/or phone.
https://users.openser.narkive.com/1aK7n8Il/sr-multihomed-kamailio-and-enable... suggests to use $sht to save the information needed, but I have no clue where in the config to put that.
Any help would be greatly appreciated.
Thanks!
-Dustin
Please check INVITE message from the sbc-ext.5060 > provider.5060 Here no Record-Route header present
06:42:30.210381 IP (tos 0x60, ttl 64, id 3430, offset 0, flags [none], proto UDP (17), length 1377) sbc-ext.5060 > provider.5060: [udp sum ok] SIP, length: 1349 INVITE sip:972xxxxxxx@xxxxxxxxxxx:5060 SIP/2.0 Via: SIP/2.0/UDP xxxxxxxxxxxxxxx;branch=z9hG4bK2eca.40a7054ddf5f0d12a23adbd4f8f501b8.0 X-OIP: 192.168.2.2 X-SBC-Check: accepted Max-Forwards: 68 From: "Dustin Marquess" sip:214xxxxxxx@sbc.xxx.net;tag=rm74ep7yFm4mm To: sip:972xxxxxxx@sbc.xxx.net Call-ID: 68dc6b49-cc32-123e-f9a0-f72abab18dff CSeq: 100863083 INVITE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 327 X-FS-Support: update_display,send_info Remote-Party-ID: "Dustin Marquess" sip:214xxxxxxx@xxx.net;party=calling;screen=yes;privacy=off X-Group-SBC: 1001 X-Routing-SBC: tr2un
Contact: sip:btpsh-685b9a1a-a0c-1@xxxxxxxxxxxxxxx
And please check "180 Ringing" from sbc-int.5060 > pbx.5080
06:42:31.265799 IP (tos 0x60, ttl 64, id 39495, offset 0, flags [none], proto UDP (17), length 994) sbc-int.5060 > pbx.5080: [udp sum ok] SIP, length: 966 SIP/2.0 180 Ringing From: "Dustin Marquess" sip:214xxxxxxx@sbc.xxx.net;tag=rm74ep7yFm4mm To: sip:972xxxxxxx@sbc.xxx.net;tag=gK0aa84cd5 Call-ID: 68dc6b49-cc32-123e-f9a0-f72abab18dff CSeq: 100863083 INVITE Contact: sip:xxxxxxxxxxx:5060;did=19d.2a212d95;transport=udp;alias=xxxxxxxxxxx~5060~1 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,OPTIONS l: 220 Content-Disposition: session; handling=required c: application/sdp Via: SIP/2.0/UDP 192.168.2.2:5080;received=192.168.2.2;rport=5080;branch=z9hG4bKrS9aSHDy40v8S Record-Route: sip:xxxxxxxxxxxxxxx;r2=on;lr;ftag=rm74ep7yFm4mm;proxy_media=yes;nat=yes,sip:192.168.2.3;r2=on;lr;ftag=rm74ep7yFm4mm;proxy_media=yes;nat=yes P-SR-XBranch: z9hG4bK2eca.40a7054ddf5f0d12a23adbd4f8f501b8.0
Here is added Record-Route headers but this headers is not present on the same message from the provider.
Looks like "record_route()" called on reply route, but this is not possible according function decscription
https://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.f.record_rout...
You need:
1) to make sure record_route() called during INVITE processing;
2) try find reason why Record-Route headers added durirng "180 RInging" processing.
On Wed, Jun 25, 2025 at 10:01 AM Dustin Marquess dmarquess@gmail.com wrote:
John,
This reply probably isn't going to be helpful, since I obviously misread your entire email. For some reason I was thinking enable_double_rr, so I set:
modparam("rr", "enable_double_rr", 2)
And tried again, same issue.
Sergey asked for the SIP messages. I didn't want to spam the list, so I put them at https://fdf.net/calls.txt
It's late now, I'll try record_route_preset in the morning and try again. 🤦
Thanks! -Dustin
On Tue, Jun 24, 2025 at 5:17 AM Who AmI myfriendjohn1@gmail.com wrote:
Just to add here, double rr with putting the interface details in them using the record_route_preset ( https://www.kamailio.org/docs/modules/devel/modules/rr.html#rr.f.record_rout...) might be the way to go here.
This will allow the BYE to have the correct routes to get back out hopefully.
So on initial invite from private: in rr header 1st rr is from public, 2nd rr is private and on initial invite from public: in rr header 1st rr is from private, 2nd rr is public
This orders the rr headers so on the way back they come in correctly if that makes sense.
I do this at the initial Invite and it works fine for me, but your mileage may vary.
Hope this helps,
John.
On Tue, 24 Jun 2025 at 10:25, Sergei Safarov via sr-users < sr-users@lists.kamailio.org> wrote:
Will be fine to check SIP messages. Here is important Route, Record-Route and contact header values.
Sergey.
On Mon, 2025-06-23 at 20:24 -0500, Dustin Marquess via sr-users wrote:
All,
I've tried searching and have found similar posts before, but I have yet to find a real 'solution'.
I'm running Kamailio 6.0.1 & RTPEngine 13.3.1.7 (on a Linux VM) as a "SBC" (at home) using the config from:
https://github.com/voiceboys/sbcOS/blob/version_2.0/var/www/html/alpine/conf...
Except that I've added the following:
mhomed=1 modparam("rr", "force_send_socket", 1)
The SBC is dual-homed: one interface on the local LAN and another interface directly on the Internet/WAN (real public IP).
Everything works fine, EXCEPT if the remote end (the side on the WAN/Internet) hangs up first. If they do, the BYE never makes it past Kamailio to the local PBX and/or phone.
https://users.openser.narkive.com/1aK7n8Il/sr-multihomed-kamailio-and-enable... suggests to use $sht to save the information needed, but I have no clue where in the config to put that.
Any help would be greatly appreciated.
Thanks!
-Dustin
-
Dustin Marquess -
Sergei Safarov -
Sergey Safarov
-
Who AmI