Call problems in baresip with freeSwitch

[Tung Giang Le]

I have a little problem in making call in baresip with freeswitch

I want to make call at least between 2 users on 2 different PC in the same network with successful registration on FS. And I also have no ideal how to make call between different network.

I have 2 PC both have baresip v1.1.0 and freeswitch 1.10.3 but in this scenario, I run baresip in both PC but only freeswitch in 1 PC and I only change config in this PC

1. If I run FS first before running baresip (uncomment sip_listen in .baresip/config) then I will encounter this error in baresip:
    tcp: sock_bind: bind: Address already in use (af=2, 192.168.11.107:5060)
    ua: SIP Transport failed: Address already in use
    ua: init failed (Address already in use)

2.    If I run FS first before running baresip (comment sip_listen in .baresip/config) then the Registraion will success but I can not received call from the other PC.

3.    If I run baresip(uncomment sip_listen) before running FS then I will be able to receive call in baresip but the registration went error:

    reg: sip:2017...@0.0.0.0 (prio 0): 501 Not Implemented (baresip v1.1.0 (x86_64/linux))

Do you guys know how to resolve these problems?

Thanks

[Christian Spielberger]

Normally you would have FS on a server in another network.

Anyway, for your registered accounts you don't need a sip_listen config. You should comment it out.

The registration should be done ideally with DNS resolution. This will automatically choose a working transport protocol. In your case, if you specify the IP address in the account file you will need also the Parameter ";transport=tcp". Because I guess that error 501 means that FS does not like UDP registrations.

See in docs/examples/accounts how to specify the transport parameter!

r,chris.v