[ASTPPCOM-1321] NATing issue with external profile Created: 22/Nov/22 Updated: 05/Sep/23 |
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| Status: | In Progress |
| Project: | ASTPP Community |
| Component/s: | None |
| Affects Version/s: | None |
| Fix Version/s: | v5.0 |
| Type: | Task | Priority: | Low |
| Reporter: | Matt Scanzano | Assignee: | Ashish Gohil |
| Resolution: | Unresolved | Votes: | 0 |
| Labels: | None | ||
| Remaining Estimate: | Not Specified | ||
| Time Spent: | 21h 20m | ||
| Original Estimate: | Not Specified | ||
| Attachments: |
Edit-SIP-Profile-default-Inextrix-Technologies-Pvt-Ltd-.png
Edit-SIP-Profile-Inextrix-Technologies-Pvt-Ltd-.png
Kazam_screenrecord_00001.mp4
Screenshot from 2023-08-19 18-40-46.png
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| Sprint: | 2023-ENT-DEV-V70-W29 |
| Edition: | Enterprise |
| Description |
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Hello,
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.
What would be the proper way to configure the SIP profile when NATing?
Thank you,
Matt |
| Comments |
| Comment by Matt Scanzano [ 25/Nov/22 ] |
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Any update on this? |
| Comment by Ashish Gohil [ 03/Aug/23 ] |
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This issue is resolved, changes done as per Nating issue with new created SIP profiles PR |
| Comment by Prashant Kumar [ 14/Aug/23 ] |
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verified in latest source community v6.0 still not working, see fs_cli log or attached video :- PSTN Call issue :- https://pastebin.freeswitch.org/view/814b41db SIP2SIP Call issue :- https://pastebin.freeswitch.org/view/b3069dc5 |
| Comment by Ashish Gohil [ 19/Aug/23 ] |
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Hello Prashant Kumar, as we have discussed, i have done new PR for the same, please ignore above PR PR in V6 https://github.com/iNextrix/ASTPP/pull/710 Please check and let me know if anything. |
| Comment by Prashant Kumar [ 19/Aug/23 ] |
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Verified in latest source community v6.0 as per verification, SIP2SIP call and PSTN working fine as expected, when registered SIP devices with Newly created SIP Profile. 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 [WRONG_CALL_STATE] See SS :- |