#536 configured STUN disallows register on inner pbx

[Anonymous]

Originally created by: YNAU...@gmail.com

What steps will reproduce the problem?
1. configured one account for use with my company's inner ip pbx and one for some sip provider
2. set up STUN 'cuz while making call via sip provider I hear no voice
3. trying to make call via inner pbx while connected over wifi to local network

What is the expected output? What do you see instead?
expected - normal call
get - cannot register on pbx. in pbx's log - attempt to work over external ip of the company's router.
more: if stun is disabled in settings nothing changes - csipsimple tries to work using external ip.

What version of the product are you using? On what operating system?
market version 00-16 [r410] - latest to the moment. there were no such troubles with previous versions.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Feature request already requested please read issue 345.

And I think that you get the problem previously ... there is no reason for not have this problem.
Stun is a centralized setting in pjsip (the sip stack csipsimple use). So setting stun unfortunately affect all accounts (read issue 345).

A workaround could be to add a dhcp lease on your phone on your Company network and use Force contact in expert setting of your company to force the use of a given IP address (the one of your dhcp lease).

> more: if stun is disabled in settings nothing changes - csipsimple tries to work using external ip
As pjsip use in addition of the stun feature another feature to detect public IP it's possible that you get globably another public IP if your sip provider is before your company account in the list (public account register => pjsip detect public ip and use it for all accounts to optimize things).

Mergedinto: 345
Status: Duplicate

[Anonymous]

Originally posted by: YNAU...@gmail.com

thanx for assistance. but is it normal that if stun is enabled i cannot use local account? is there any workaround for it?

[Anonymous]

Originally posted by: YNAU...@gmail.com

i need both at a time.

[Anonymous]

Originally posted by: r3gis...@gmail.com

I understand
Read my comment 2 on the issue 345.
https://code.google.com/p/csipsimple/issues/detail?id=345#c2

For now that's the state of the native sip stack we are relying on.

It's opensource and can be contributed by anyone. So if I get time I'll take time to fix that (workaround that). For now it's not a priority for me, but the issue is open in CSipSimple project (issue 345) so it's not a wontfix. Just a "for now I can't do that easily, if pjsip guys add the feature, I'll have it too, else we have to workaround" (and for now the only workaround I can find is what I said previously in this issue (and in comment 2 of my post on issue 345) ).
If any contributor to CSipSimple or pjsip add the feature, I'll integrate that of course :)