Newbie: one extension can call a second extension but not vice versa

I have installed and updated the Freepbx to the latest version.
I created two extensions: 200 and 201
Both extensions and the freepbx server are on the same network (lan) without any subnet or routing.
200 can call 201
201 cannot call 200: I get the message 503 service unavailable.
200 can call itself
201 cannot call itself it gets the message: Call failed: 486 busy.
Both extensions use Eyebeam softphones.
I have tried it with X-Lite as well.

I have restarted the linux box, just in case, however I get the same behavior as above.

Any idea where I should be looking? dial plans, extensions etc…

200 can call 201:
ringing is the signalling part, did u check speech path as well? (I mean, did u answer 201?)

201 cannot call 200:
*check if there is a codec missing from 200
*check if u have DND (do not disturb) on 200

200 can call itself:
probably, 200 has cw (call waiting active)

201 cannot call itself it gets the message: Call failed: 486 busy:
yes, if there is no cw active

anyway,
1.post configuration from asterisk for 200 and 201
2.also, post configuration from softphones (both)
3.post SIP traces from 200 --> 201 and 201–>200

200 can call 201:
ringing is the signaling part, did u check speech path as well? (I mean, did u answer 201?)

Yes.

201 cannot call 200:
*check if there is a codec missing from 200

How can I check the codec and where? in the SIP client or Freepbx admin? or Astersisk CLI?

*check if u have DND (do not disturb) on 200

OK I will (as I do not have access to it right now)

200 can call itself:
probably, 200 has cw (call waiting active)
201 cannot call itself it gets the message: Call failed: 486 busy:
yes, if there is no cw active

OK

1.post configuration from asterisk for 200 and 201

You mean the sip.conf or extensions.conf etc…

2.also, post configuration from softphones (both)

you mean the domain of the Asterisk, extension number, etc… in the preferences/options section of the SIP client or is there any special file I should look for?

3.post SIP traces from 200 --> 201 and 201–>200

Where do I check these traces?

Thanks for all the help

DND :slight_smile:

Thanks a lot, the other SIP phone DND was on…