partysip-dev
[Top][All Lists]
Advanced

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

Re: [Partysip-dev] Partysip and Cisco ATA 186


From: Jason
Subject: Re: [Partysip-dev] Partysip and Cisco ATA 186
Date: Mon, 9 Sep 2002 19:43:59 +0800

Hi Binand,

    When the partysip receives an invite request, it sends a 100 (trying)
response to the caller and forward the request to the callee.
    But it doesn't forward the follow up 1XX resoponse, received from the
callee, to the caller.
    Therefore, the caller will not receive the 180 (ringing) response sent
form the callee.
    You can modify some codes in sfp.c and psp_core5.c to let the partysip
forward  upstream follow up 1XX response.

Jason

Regards

----- Original Message -----
From: "Binand Raj S." <address@hidden>
To: <address@hidden>
Sent: Monday, September 09, 2002 6:44 PM
Subject: [Partysip-dev] Partysip and Cisco ATA 186


> Hi,
>
> I have partysip serving for a few (5) Cisco ATA 186's. The setup works
> well, except for one small annoyance - the calling party does not hear a
> ring tone when the phone is indeed ringing at the called end. Can anyone
> tell me why this is happening?
>
> I am sort of new to partysip/osip/cisco ATAs, and this issue has stumped
> me so far.
>
> This feature was working when we used these phones with a third party SIP
> provider.
>
> Binand
>
>
>
> _______________________________________________
> Partysip-dev mailing list
> address@hidden
> http://mail.freesoftware.fsf.org/mailman/listinfo/partysip-dev






reply via email to

[Prev in Thread] Current Thread [Next in Thread]