[asterisk-users] Qualify renders all SIP peers unreachable

Jaswinder Singh vicky.r at gmail.com
Thu Jun 14 13:42:21 CDT 2007


What does "sip show peers" output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes

On 14/06/07, randulo <spamsucks2005 at gmail.com> wrote:
>
> I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
> All SIP peers are working properly to place or receive calls.
> Any SIP peer or friend whether NATted or not will become UNREACHABLE
> if qualify=yes.
>
> I have identical peers on the other asterisk 1.2.16 production server.
> In fact, two of the phones (linksys 941 and Polycom ip500) are using
> one line for each asterisk. The 1.2 one works normally, the 1.4 does
> not.
>
> The sip confgs from "sip show settings" are identical on the two servers.
>
> The sip.conf peer entries were moved over exactly.
>
> Ports 5060 to 5065 are forwarded to the asterisk server.
>
> Looking at sip debug, I notice a few differences:
>
> REGISTER from phone:
>
> "Authorization: Digest username="Poly", realm="asterisk",..."
>
> does not show on the 1.4 server.
>
> Trying (sent by *):
>
> "Supported: replaces"
>
> The Via lines are the same (internal ip addresses) on both servers,
> but there is a "Sending to 192.168..." on the 1.2 message where there
> is none on the 1.4.
>
> What is "supported: replaces" ?
>
> What config setting generates the "Authorization: Digest..." message ?
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070614/903eba51/attachment.htm


More information about the asterisk-users mailing list