Hi all,
I have configured Polycom Trio SIP Phone as SIP 3party on the CS1K system and it is registered correclty:
Incoming calls (From CS1K Unistim phone to Polycom SIP Phone) are working fine
When we call from SIP Phone to Unistim phone, I got busy tone, and on the traces I can see 407 authentification required returned by CS1K followed by 403 Forbidden
When we redial from missed call of the SIP Phone (Polycom) to call Unistim phone, it works fine !!
In my investigation, when I compare bad and good calls, I can see that the good calls (redial from missed calls list) sent the INVITE as follow:
"sip:1000dialsrc=calllistBphone-contextdialsrc=calllistDcdp.udp@mydomain"
Bad call (normal outbound call) with INVITE like: "sip:1000@mydomain;user=phone"
I guess that the CS1K don't accept INVITE format without "phone-context=cdp.udp"
Please any idea of how to put adaptation or something on the CS1K side ?
Thank you.
I have configured Polycom Trio SIP Phone as SIP 3party on the CS1K system and it is registered correclty:
Incoming calls (From CS1K Unistim phone to Polycom SIP Phone) are working fine
When we call from SIP Phone to Unistim phone, I got busy tone, and on the traces I can see 407 authentification required returned by CS1K followed by 403 Forbidden
When we redial from missed call of the SIP Phone (Polycom) to call Unistim phone, it works fine !!
In my investigation, when I compare bad and good calls, I can see that the good calls (redial from missed calls list) sent the INVITE as follow:
"sip:1000dialsrc=calllistBphone-contextdialsrc=calllistDcdp.udp@mydomain"
Bad call (normal outbound call) with INVITE like: "sip:1000@mydomain;user=phone"
I guess that the CS1K don't accept INVITE format without "phone-context=cdp.udp"
Please any idea of how to put adaptation or something on the CS1K side ?
Thank you.