SURE! This is the Sofia Internal side... http://pastebin.com/xwBjzMCX This is Javascript console... (JsSIP -> OverSIP) http://pastebin.com/XmXRt0nV Thanks. 2015-05-19 18:15 GMT-04:30 Brian West <brian@freeswitch.org>:
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 +58424 291 4561 BB #79A8AFA2 @VMCibersys
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-- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys