Forgot to say running with Asterisk 1.2.10 mainline code on RedHat FC5 box.
Other Cisco 7960 (SIP 7.5) and 7912 phones (SIP ver 1.3.1?) around the house
work as expected including MWI...
----- Original Message -----
From: "Michael J. Tubby G8TIC" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Cc: <[EMAIL PROTECTED]>
Sent: Monday, August 14, 2006 4:09 PM
Subject: [asterisk-users] Re: Cisco 7970 MWI not working (Was: Problem
withCisco7970 SIP load / call transfer)
Juha,
I am running the same version of Cisco 7970 SIP firmware and having the
same problem with periodic 400 "Bad Request" responses from it when
Asterisk sends MWI updates for a voicemail box...
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
-- Got SIP response 400 "Bad Request" back from 192.168.144.187
so, Cisco have changed what they are expecing in the SIP headers for the
MWI to work...?
My hunch is that they are wanting an @<ipaddress of server> somewhere in
the notification as this would be consistent with the other changes that
they have made for resillience when the phone talks to more than one
CallManager 5.x server in SIP mode -- hence why people are commenting on
the @<ip address> turning up in caller id. Actually it makes sense if
there are to be multiple servers supporting a HA phone system because when
you have a missed call and hit the "call them back" button you probably do
need to do it in the context of the call you missed (ie. via the server
that Invited you).
That being said the MWI not working is a pain! Does anyone on the list
have the ability to capture working SIP MWI notifications from a
CallManager 5.x talking to a Cisco 7960 phone running SIP 8.0.2 using
Ethereal (or some other packet sniffing tool) so we can see what the SIP
looked like and fix (patch) Asterisk???
Regards
Mike
----- Original Message -----
From: "Juha Suhonen" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Wednesday, August 02, 2006 8:27 AM
Subject: [asterisk-users] Problem with Cisco7970 SIP load / call transfer
Hi!
I'm having an interesting problem with Cisco7970 SIP load (8.0(2)SR1) -
the phone seems to work otherwise fine, but I can't do an assisted
transfer (and the 7970 phone also doesn't seem to support the BlindXFer
option that previous models have had). Phones are connected to Asterisk
1.2.10.
What happens is this: User a calls to my phone. I press "Transfer" on the
phone, I then place another call to another extension. When this new call
is connected, pressing the "Transfer" -button again sends 2 SIP INVITE
messages (and asterisk acks them with seemingly appropriate "OK"
messages). But.. After getting the acks, phone just says "Unable to
complete transfer" and both current calls are placed on hold.
Has anybody else seen this? Any ideas on how to fix? The same
configuration works with Cisco 7960 (using some pretty ancient SIP load).
I've also thought about upgrading the phone to 8.0(3) release of the SIP
load, but atleast voip-info.org wiki states it as a "total disaster" -
can anybody confirm if it's really a disaster?
As a related note, I'm also not seeing MWI with the 7970 phone - when
Asterisk sends the MWI status message to phone, Asterisk immediatetly
barfs out -- Got SIP response 400 "Bad Request" back from xxx. Does
anybody know if this is a bug on the phone and maybe fixed on a later
image? (and is there any workaround I can enable on asterisk to overcome
this)
Also, a small UI thing - has anybody found a way to get the # -key to
directly dial the number which has been inputted and mimic the behaviour
7960s had? Our users are accustomed to keying in 123# instead of pressing
123 + "dial"..
-- juhas
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