[Freeswitch-users] JsSIP -> OverSIP -> Freeswitch

Brian West brian at freeswitch.org
Wed May 20 02:45:19 MSD 2015


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 at 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 at conference.cibersys.com>;tag=5or80k3oc9
> To: <sip:9196 at 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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-- 

*Brian West*
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