Comments on: Configuring Cisco 7975 IP Phones for SIP

By: Tyler Winfield

Tyler Winfield — Tue, 20 Jul 2010 18:13:34 +0000

Anyone having problems with their Cisco IP Phones, please see the article i've put together on getting SEVERAL different models connected to asterisk with SIP firmware.
http://minded.ca/2009-12-16/configure-cisco-ip-ph...

Regarding the TFTP ports (aka DHCP options), start here: http://www.cisco.com/en/US/products/sw/voicesw/ps...

According to this CCM 3.3 (the link on the page to the CCM 4.1 info is the same) the TFTP port (or DHCP option) is port 69, NOT 66 or 150 as many documents will indicate. If this doesn't work, the phones network settings contain a spot for "Alternate TFTP". Setting this will ignore any passed DHCP options and try to connect to the provided address on port 69 directly.

Also, best experiences to date have come from 8.4.X firmware versions of the SIP images

By: voipstore

voipstore — Tue, 20 Apr 2010 19:14:04 +0000

The Linksys phones are being phased out and are replaced with the Cisco SPA5xx series phones.
-Kerry

By: wvds-nl

wvds-nl — Tue, 20 Apr 2010 19:11:25 +0000

1. If you use SIP, then don’t use Cisco. Try linkys (wich is also cisco) instead.

2. If you really like to use Cisco (cause their phones are really decent and beautiful) then use it in combination with SCCP (chan_skinny for asterisk and mod_skinny for FreeSWITCH).

By: voipstore

voipstore — Tue, 13 Apr 2010 16:50:11 +0000

It is very difficult to troubleshoot these which is why I do not recommend using them with Asterisk. Without having access to the system its even harder to figure out why they aren’t working. Try changing the NAT and Qualify settings to see if that helps.
-Kerry

By: Alessio

Alessio — Tue, 13 Apr 2010 16:46:29 +0000

Hi all,

I have a 7975 and they are 2 days that I'm trying to connect it to my Asterisk server…
I have installed firmware SIP 8-3-3 (but also 8-5-4 for some tests).

Right now, I'm using the above configuration from skintigh, deleting the NAT entries. http://voip-info.linuxsys.com/wiki/indexb7292ddf4...
7975 is not able to register to Asterisk.

Strange behaviour: 7975 is NOT registered to Asterisk (grey icon with red cross), but is able to call other phones through Asterisk. Instead, other phones are not able to call the 7975.

During the registration, Asterisk CLI reports:

– Registered SIP '0011' at 172.16.2.21 port 5060
> Saved useragent "Cisco-CP7975G/8.3.0" for peer 0011
— Unregistered SIP '0011'

In status messages, 7975 tells:
Anonymous (a part from Error Updating Locale).

My users.conf in Asterisks sounds like this:

[0011]
fullname = console
cid_number = 0011
nat = no ; there is not NAT between phone and Asterisk
insecure = invite
type = friend
allow = all
context = console
hasmanager = no
callwaiting = yes
threewaycalling = yes
hasagent = yes
group =
host = dynamic
vmsecret =
registersip = yes
registeriax = no
;limitonpeers = yes
;incominglimit=1
;call-limit=1
secret = xxxxxxx

Any proposal??? I don't know now what to do…

Thanks in advance
Alessio

By: Yuval

Yuval — Mon, 29 Mar 2010 08:45:30 +0000

Tried to install the phone with the 8.3.2 SR1S which I took from the cisco download software area of 7975. Unlike previous versions it is written there that this file is for the callmnager. On the 7961 and others you have SIP files for "other systems".
What to do – to take this version from 7961 ? (since with this files I can see the process is running but at the end it makes reset and restart the process again and again …)

By: John Challenor

John Challenor — Sat, 27 Feb 2010 00:38:18 +0000

Now I wonder how to get the call waiting to work so the 2nd call rings into the 2nd line button instead of the "soft" line key on the display……

By: John Challenor

John Challenor — Fri, 26 Feb 2010 19:18:24 +0000

that did it!!!! Thanks!!!

By: John Challenor

John Challenor — Fri, 26 Feb 2010 00:20:07 +0000

trixbox CE

By: voipstore

voipstore — Thu, 25 Feb 2010 19:43:19 +0000

With trixbox CE, edit the extension, set NAT=no and Qualify=Yes