[asterisk-dev] Inquiry:Asterisk sip ?

[hadi motamedi]

On Sat, Jan 2, 2010 at 5:04 AM, hadi motamedi <motamedi24 at gmail.com> wrote:

> Dear All
> I sent the following message to Users mailing list but no reply . Can you
> please do me favor and take a look at the attached log and let me know what
> is my problem that causes the call to drop upon called party goes offhook ?
> Thank you in advance
>
>
>
> ---------- Forwarded message ----------
> From: hadi motamedi <motamedi24 at gmail.com>
> Date: Thu, Dec 31, 2009 at 6:40 AM
> Subject: Inquiry:Asterisk sip ?
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
>
>
> Dear All
> Please be informed that my Asterisk has sip connection to an external
> sip server but the sip outgoing call will be disconnected for some
> unknown reasons . Please find attached the debug log . Can you please
> do me favor and let me know what is the problem that causes the call
> to immediately being dropped when the called party goes offhook ?
> Thank you
>
>

Dear All
Please be informed that the problem came from "canreinvite=yes" settings .
It changed to "canreinvite=no" and the problem solved out.
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