https://jira.astppbilling.org This file is an XML representation of an issue en-us 8.13.3 813003 14-01-2021 https://jira.astppbilling.org/browse/ASTPPCOM-1321 ASTPP Community <p>Hello,</p> <p> </p> <p>I created a external profile for a NATed connection in ASTPP. I don't seem to be able to pass RTP properly as the RTP ip is still the local IP when capturing so there is never a return in audio.</p> <p> </p> <p>What would be the proper way to configure the SIP profile when NATing?</p> <p> </p> <p>Thank you,</p> <p> </p> <p>Matt</p> ASTPPCOM-1321

NATing issue with external profile

Task Low In Progress Unresolved Ashish Gohil Matt Scanzano Tue, 22 Nov 2022 07:06:50 +0530 Tue, 5 Sep 2023 12:28:22 +0530 v5.0 0 3 21h 20m <p>Any update on this?</p> <p>This issue is resolved, changes done as per Nating issue with new created SIP profiles</p> <p>PR</p> <p><a href="https://github.com/iNextrix/ASTPP/pull/704" class="external-link" target="_blank" rel="nofollow noopener">https://github.com/iNextrix/ASTPP/pull/704</a></p> <p>verified in latest source community v6.0</p> <p>still not working, </p> <p>see fs_cli log or attached video :- PSTN Call issue :-  <a href="https://pastebin.freeswitch.org/view/814b41db" class="external-link" target="_blank" rel="nofollow noopener">https://pastebin.freeswitch.org/view/814b41db</a> </p> <p>SIP2SIP Call issue :- <a href="https://pastebin.freeswitch.org/view/b3069dc5" class="external-link" target="_blank" rel="nofollow noopener">https://pastebin.freeswitch.org/view/b3069dc5</a> </p> <p>Hello <a href="https://jira.astppbilling.org/secure/ViewProfile.jspa?name=prashant.kumar" class="user-hover" rel="prashant.kumar">Prashant Kumar</a>, as we have discussed, i have done new PR for the same, please ignore above PR</p> <p>PR in V6</p> <p><a href="https://github.com/iNextrix/ASTPP/pull/710" class="external-link" target="_blank" rel="nofollow noopener">https://github.com/iNextrix/ASTPP/pull/710</a></p> <p>Please check and let me know if anything.</p> <p>Verified in latest source community v6.0</p> <p>as per verification, SIP2SIP call and PSTN working fine as expected, when registered SIP devices with Newly created SIP Profile.</p> <p>But when dialed DID number from other server, and try to receive the call on DID Number, then it didn't it displayed error as <b><span class="error">&#91;WRONG_CALL_STATE&#93;</span></b> </p> <p>See SS :- </p> ASTPP Edition Development Edition RCA Rank 0|i04p4v: Sprint 2023-ENT-DEV-V70-W29 Support Type