The whole sip exchange would need to be seen for me to tell, it could be a bug, it could be a config issue... not exactly sure yet.  Can you paste bin it?

Thanks,

On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina@cibersys.com> wrote:

Hi guys!

Im having some problems while trying to connect to a Freswitch PBX using OverSip and JsSIP.

Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I always get a 422 error.

Softphone -> WebRTC ext calls OK
WebRTC -> Softphone fails
WebRTC -> ECHO TEST call on freeswitch also fails

Can somebody help find out what this could be?

SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
From: "test" <sip:1002@conference.cibersys.com>;tag=5or80k3oc9
To: <sip:9196@conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
Call-ID: 4norlmjdkqok6smt0vd4
CSeq: 9416 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Min-Se: 120
Content-Length: 0

BTW... already asked @oversip guys... But wanted to try this in here just in case! =)

Thanks.

--

Víctor E. Medina M.

Platform Architect / Chief Infrastructure

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BB #79A8AFA2
@VMCibersys

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