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Hi Bogdan.</div>
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Thanks for the help, <span style="font-family: "Courier New", Arial, Helvetica, sans-serif;">
script_trace</span> helps a lot. However, one problem appears to be our Asterisk sending BYEs to the OpenSIPS management IP, rather than the service IP - the other is no BYEs appearing from our upstream provider, so there may yet be some NAT fiddling in the
network.</div>
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This is a tcpdump of a short test call from the OpenSIPS host, repeating lines have been cropped.</div>
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18:37:20.965341 IP <SUPPLIER_IP_1>.5060 > <OSIPS_SVC_IP>.5060: SIP: INVITE sip:<OUR_DDI>@<OSIPS_SVC_IP> SIP/2.0</div>
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18:37:20.976766 IP <OSIPS_SVC_IP>.5060 > <SUPPLIER_IP_1>.5060: SIP: SIP/2.0 100 Giving it a try</div>
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18:37:20.976914 IP <OSIPS_SVC_IP>.5060 > <AST_SVC_IP>.5060: SIP: INVITE sip:<INTERNAL_NO>@<AST_SVC_IP> SIP/2.0</div>
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18:37:20.980867 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 100 Trying</div>
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18:37:20.983652 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 180 Ringing</div>
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18:37:20.983956 IP <OSIPS_SVC_IP>.5060 > <SUPPLIER_IP_1>.5060: SIP: SIP/2.0 180 Ringing</div>
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...</div>
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18:37:24.494830 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 200 OK</div>
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18:37:24.512790 IP <OSIPS_SVC_IP>.5060 > <SUPPLIER_IP_1>.5060: SIP: SIP/2.0 200 OK</div>
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...</div>
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18:37:27.994889 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 200 OK</div>
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18:37:27.995144 IP <OSIPS_SVC_IP>.5060 > <SUPPLIER_IP_1>.5060: SIP: SIP/2.0 200 OK</div>
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18:37:31.995201 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 200 OK</div>
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...</div>
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18:37:43.995343 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 200 OK <-- Hangup</div>
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...</div>
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18:37:55.995731 IP <AST_SVC_IP>.5060 > <OSIPS_SVC_IP>.5060: SIP: SIP/2.0 200 OK</div>
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18:37:56.495480 IP <AST_SVC_IP>.5060 > <OSIPS_MGMT_IP>.5060: SIP: BYE sip:<CALLERS_DDI>@<SUPPLIER_IP_2>:5060 SIP/2.0</div>
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18:37:56.994867 IP <AST_SVC_IP>.5060 > <OSIPS_MGMT_IP>.5060: SIP: BYE sip:<CALLERS_DDI>@<SUPPLIER_IP_2>:5060 SIP/2.0</div>
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...</div>
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18:38:23.994375 IP <AST_SVC_IP>.5060 > <OSIPS_MGMT_IP>.5060: SIP: BYE sip:<CALLERS_DDI>@<SUPPLIER_IP_2>:5060 SIP/2.0</div>
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18:38:27.995061 IP <AST_SVC_IP>.5060 > <OSIPS_MGMT_IP>.5060: SIP: BYE sip:<CALLERS_DDI>@<SUPPLIER_IP_2>:5060 SIP/2.0</div>
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and the headers from an INVITE packet</div>
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18:44:04.428866 IP <OSIPS_SVC_IP>.5060 > <AST_SVC_IP>.5060: SIP: INVITE sip:<INTERNAL_NO>@<AST_SVC_IP> SIP/2.0</div>
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E..Jb-@.@...%<n.%<n......6+.INVITE sip:<INTERNAL_NO>@<AST_SVC_IP> SIP/2.0</div>
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Record-Route: <sip:<OSIPS_MGMT_IP>:5060;lr;did=e9f.c2eba4c6></div>
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Record-Route: <sip:<SUPPLIER_IP_3>;lr;ftag=gK0e701262;did=e9f.85477a97></div>
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Via: SIP/2.0/UDP <OSIPS_MGMT_IP>:5060;branch=z9hG4bK421c.1ab0cbb5.0</div>
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Via: SIP/2.0/UDP <SUPPLIER_IP_3>:5060;branch=z9hG4bK421c.ffa73133.0</div>
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Via: SIP/2.0/UDP <SUPPLIER_IP_2>:5060;branch=z9hG4bK0eB8733317d12c9fc4b</div>
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From: <sip:<CALLERS_DDI>@<SUPPLIER_IP_4>;user=phone>;tag=gK0e701262</div>
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To: <sip:<OUR_DDI>@<SUPPLIER_IP_3>;user=phone></div>
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Call-ID: 355393690_100431806@<SUPPLIER_IP_2></div>
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CSeq: 493596 INVITE</div>
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Max-Forwards: 68</div>
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Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH</div>
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Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed</div>
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Contact: <sip:<CALLERS_DDI>4@<SUPPLIER_IP_2>:5060></div>
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P-Asserted-Identity: <sip:<CALLERS_DDI>@<SUPPLIER_IP_2>:5060;user=phone></div>
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...</div>
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In the configuration I've set:</div>
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socket=udp:<OSIPS_SVC_IP>:5060 as <OSIPS_MGMT_IP>:5060</div>
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If a call is allowed through the INVITE path I previously posted, a routing block with a variation in the
<span style="font-family: "Courier New", Arial, Helvetica, sans-serif;">$ru</span> value will send it to the appropriate internal host.</div>
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route[ALLOW]{</div>
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if ($ru =~ "sip:111[0-9]@*"){</div>
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$avp(routegroup) = $(avp(site){s.int});</div>
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do_routing($avp(routegroup));</div>
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route(REJECT);</div>
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exit;</div>
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}</div>
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...</div>
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Does it look like I've misconfigured something, or is Asterisk skipping a host & sending the BYEs to the host one beyond its peer?</div>
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Asterisk has no knowledge of the OpenSIPS management IPs in its configuration.</div>
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Thanks again,</div>
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Al.</div>
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<b>From: </b>Bogdan-Andrei Iancu <bogdan@opensips.org><br>
<b>Date: </b>Monday, 2 February 2026 at 9:31 am<br>
<b>To: </b>OpenSIPS users mailling list <users@lists.opensips.org>, Alistair Cleminson <alistair.cleminson@wifinity.co.uk><br>
<b>Subject: </b>Re: [OpenSIPS-Users] Problems passing BYE signal in v3.7<br>
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<div class="ms-outlook-mobile-reference-message skipProofing">Hi Alistiar,<br>
<br>
The BYE is a sequential/indialog request, so you need to check your script on how these are handled. Usually they are routed via record_route() (if the routing mechanism is used in your cfg). See
<a href="https://github.com/OpenSIPS/opensips/blob/3.6/etc/opensips.cfg#L110" class="moz-txt-link-freetext" originalsrc="https://github.com/OpenSIPS/opensips/blob/3.6/etc/opensips.cfg#L110" data-outlook-id="b5b06d33-670f-42e6-a8c5-0c5d4d9dbfad">
https://github.com/OpenSIPS/opensips/blob/3.6/etc/opensips.cfg#L110</a><br>
<br>
IF you do not understand you cfg flow, maybe you can add this script tracing function on top of your cfg, so you can see how the execution goes step by step via the cfg:
<a href="https://www.opensips.org/Documentation/Script-CoreFunctions-3-6#script_trace" class="moz-txt-link-freetext" originalsrc="https://www.opensips.org/Documentation/Script-CoreFunctions-3-6#script_trace" data-outlook-id="ae7ea6cd-0d42-45e0-9553-62122bd2bc27">
https://www.opensips.org/Documentation/Script-CoreFunctions-3-6#script_trace</a><br>
<br>
Regards,</div>
<pre><div class="moz-signature">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="https://www.opensips-solutions.com/" class="moz-txt-link-freetext" originalsrc="https://www.opensips-solutions.com/" data-outlook-id="995035f6-2725-4e8c-a409-c4d15423755c">https://www.opensips-solutions.com</a>
<a href="https://www.siphub.com/" class="moz-txt-link-freetext" originalsrc="https://www.siphub.com/" data-outlook-id="e5695272-c949-4f19-a5c5-0c9f96175436">https://www.siphub.com</a></div></pre>
<div class="moz-cite-prefix">On 30.01.2026 18:54, Alistair Cleminson wrote:</div>
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Thanks for the tip, it turns out I'd completely misunderstood the process. This configuration doesn't generate the errors & passes calls:</div>
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<snip></div>
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<b>From: </b>johan de clercq <a href="mailto:johan@democon.be" class="moz-txt-link-rfc2396E" data-outlook-id="fd931c9d-e19e-4119-b51a-cd498d234445">
<johan@democon.be></a><br>
<b>Date: </b>Thursday, 29 January 2026 at 9:41 am<br>
<b>To: </b>Alistair Cleminson <a href="mailto:alistair.cleminson@wifinity.co.uk" class="moz-txt-link-rfc2396E" data-outlook-id="9064c5f8-fe11-49d8-8773-61c15fc91691">
<alistair.cleminson@wifinity.co.uk></a><br>
<b>Subject: </b>Re: [OpenSIPS-Users] Problems passing BYE signal in v3.7<br>
<br>
</div>
maybe changing the 2 vars to avp's will fix the issue.
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<div class="moz-cite-prefix">On 1/27/26 17:27, Alistair Cleminson wrote:</div>
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<span style="text-transform: none;">Hi,</span></div>
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I've nearly completed the process of upgrading our ancient v1.7 deployment to v3.6. The new deployment is installed on Ubuntu 24.04 from the OpenSIPS repos. These servers are our inbound hand-off from our upstream supplier, they take a call, inspect the called
number element against the database, and then pass it on to the target system with the internal reference or decline the call.</div>
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<snip></div>
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