On 02/14/2011 12:04 PM, James Miller wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the
ring type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1 ring_att1.pcm
snip
Do you actually have
On Mon, Feb 14, 2011 at 10:31 AM, James Miller paramedi...@gmail.com wrote:
I did that and this is what I got when I tried to play the 24 ringtone:
13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency
ring_emergency.pcm octet
That line should read something like:
blah..
Yes, nothing changed EXCEPT for the software image the phone pulled down.
All of the files are still in the exact same locations with the exact same
names as they had in 8.9. I'm at a loss as to what's causing this issue and
so apparently is Cisco given they have yet to respond to my follow up
Problem has been resolved with the assistance of Jonathan. Appears to be an
issue with my text editors not properly tabbing the file correctly.
Regards.
I see blindness, not as a disability, but more of an ability. And Sight
actually, more of a disability because some people with sight tend to
Good Day everyone,
Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by
Cisco, however now the phone does not and will not read the RINGLIST.dat
file. I've tried rebooting the phone, tried resetting the phone back to
factory, have deleted the RINGLIST.dat file and
Better to report a BUG to cisco.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller
Sent: Monday, February 14, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22
the future's of the free world.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif
Sent: Monday, February 14, 2011 8:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Cisco 7960
On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote:
Good Day everyone,
Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by
Cisco, however now the phone does not and will not read the RINGLIST.dat
file. I’ve tried rebooting the phone, tried
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan
Thurman
Sent: Monday, February 14, 2011 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat
error
On Mon, Feb 14, 2011 at 9:04 AM, James Miller paramedi...@gmail.com wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the ring
type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1
I did that and this is what I got when I tried to play the 24 ringtone:
13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency
ring_emergency.pcm octet
In the ringlist.dat file in the first column I typed the display name then
hit the tab key. Now on some it only moved a
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4
and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware.
The Asterisk setup relies heavily on queues with dynamic agents. The problem I
am having is that on SOME (but not all) the Cisco phones,
Check your dialplan.xml file that the affected phones are loading.
Thanks,
--Warren Selby
On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com
wrote:
I recently inherited an Asterisk system (PBX in a Flash, based on
Asterisk 1.4 and FreePBX). The phones are mostly Cisco
!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Sunday, July 25, 2010 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 phone can't
-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Sunday, July 25, 2010 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue
Sent: Sunday, July 25, 2010 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue
It may be pulling a tftp server from dhcp, or it may just have an old config.
Do all the phones (even the ones that work properly) use
On Sun, Jul 25, 2010 at 9:52 AM, Kevin Keane subscript...@kkeane.comwrote:
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk
1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP
firmware.
The Asterisk setup relies heavily on queues with dynamic
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mstults tds.net
Sent: Sunday, July 25, 2010 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue
On Sun
Alyed wrote:
From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
If you turn on *qualify* in the configuration of a SIP device in
sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf,
asterisk will send a SIP
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable, and the phones lose their registration.
The only way to make the phones recover is to clear the NAT
translation tables for the phones on the PIX
On Mon, Mar 29, 2010 at 12:25 AM, Troy Davis t...@yort.com wrote:
sip fixup is enabled on the PIX
Try disabling the sip fixup on the PIX and see if that helps. You may have
to adjust the configs on the phones themselves when you do this.
--
Thanks,
--Warren Selby
http://www.selbytech.com
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable, and the phones lose their registration.
The only way to make the phones recover is to clear the NAT
translation tables for the phones on the PIX (clear
From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
If you turn on *qualify* in the configuration of a SIP device in
sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf,
Asterisk will send a SIP
OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand
regularly
Hello Everyone,
I was trying to use md5secrets with the Cisco 7960 phone, however I failed.
I did some experiments and found out the Cisco 7960 can accept passwords up
to 31 characters, while the md5sum generates passwords which are 32 chars
long.
Does anyone has the solution or faced this
Hi
You could also do it with one extension but set the call limit for the
extension in the sip.conf to something like
call-limit=3
Which would allow 3 concurrent calls to the one extension
Ish
Jimmy Ezell wrote:
Thanks for the help, I really appreciate the feedback.
I tried ringing them
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of D Tucny
Sent: Tuesday, August 11, 2009 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 Multiline phone
Thanks for the help, I really appreciate the feedback.
I tried ringing them all at the same time as you suggested:
exten =
workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomm
ing4SIP/incomming5)
but it does very strange stuff:
- I have to push the extension button twice
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell
Sent: Tuesday, August 11, 2009 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960 Multiline phone
Thanks for the help, I really appreciate the feedback.
I tried ringing
Sent: Tuesday, August 11, 2009 12:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Cisco 7960 Multiline phone
Jimmy,
To clarify, you want to configure the phones like this where p
means phone
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming extension 3
Front Desk Phone line 4 - incoming extension 4
Front Desk Phone line 5 - incoming extension 5
: [asterisk-users] Cisco 7960 Multiline phone
Jimmy,
To clarify, you want to configure the phones like this where p means phone
and l means logical line:
Phone 1:
P1l1
P1l2
P1l3
Phone 2:
P2l1
P2l2
P2l3
Phone 3:
P3l1
P3l2
P3l3
It sounds like (and looks like) you’re
On Tue, Aug 11, 2009 at 5:12 PM, Jimmy Ezell jez...@hmhca.com wrote:
Sorry for not being real clear.
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming
Thanks Guys,
I managed to get it working the problem was NAT;
in the sip.conf
[general]
nat=yes
however in the SIPMAC.cnf there was nothing about NAT.
It took me a while to spot it since both asterisk and phone were in
same network and I did not think about NAT.
Solutions:
1) add in sip.conf
Hello Mark,
I managed to make it work - see my previous post
Since you have those phones - does:
voip_control_port: 5060
start_media_port: 1
end_media_port: 10050
works in your case? I tried to put those in SIPDefault but looks like
the phone ignores those and always says:
start media
Are there any other phones registered, or is it just this phone that is
having issues? The first thing that I see is the qualify=200 line, and I
have not had good experience with Cisco devices and any qualify setting. I
would try leaving that out. I also have double quotes around the line1_*
Dear All,
I'm trying to configure my new phone Cisco 7960 to work with asterisk.
I followed
http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html
and I got into the point where I can see on the the display line indication
showing
55 phone icon with x so it looks like the
As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device,
Discussion; Darryl
Dunkin
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
I tried increasing the value and even set it to never and added the
qualify line but that did not help. Do I need to poke any holes in the
firewall on the nat device for the udp traffic to stay
Message-
From: Stephen Reese [mailto:[EMAIL PROTECTED]
Sent: Saturday, October 18, 2008 14:41
To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl
Dunkin
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
I tried increasing the value and even set
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone
.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco
] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another
] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne
Sent: Thursday, October 09, 2008 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
Hi All,
I'm thinking of creating a new asterisk server using the latest 1.4
stable release to replace
it's a nice setup.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne
Sent: Thursday, October 09, 2008 6:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
Hi All
Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
Hi All,
I'm thinking of creating a new asterisk server using the latest 1.4
stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its
been a while!).
My only concern - my phones
On 21:28, Fri 10 Oct 08, Wayne wrote:
Thanks both,
The only thing I have a little concern over is that 1.6 is that its
still a development release (if I understand things correctly).
No, 1.6.0 has been released. This is indeed the first public 'final'
release of the 1.6 series. But it's
Hi All,
I'm thinking of creating a new asterisk server using the latest 1.4
stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its
been a while!).
My only concern - my phones are Cisco 7960's (with sccp firmware 7.2
loaded) and to support them better, I remember compiling in a
On Fri, Aug 15, 2008 at 12:27:16PM -0500, [EMAIL PROTECTED] wrote:
Hello,
I have recently setup my first PBX and am wondering if there might be a
way to send audible notification to the cisco 7960 phone when a call is
put on hold. We lost a call due to a customer being on hold and
An educated guess is:
reverse the SIP trunk buttons, so the preferred provider is the top
button, and voila, your speed dial going to the first trunk is now
what you want.
On Wed, Aug 13, 2008 at 7:44 PM, Shawn L [EMAIL PROTECTED] wrote:
This one is a little off-topic, it's more about the phone
Hello,
I have recently setup my first PBX and am wondering if there might be a
way to send audible notification to the cisco 7960 phone when a call is
put on hold. We lost a call due to a customer being on hold and
forgotten about (yikes). Is there a way to get the phone to beep or ring
down the
This one is a little off-topic, it's more about the phone than asterisk
itself.
I have a cisco 7960 configured with 2 lines to 2 different sip providers
(cant get
asterisk to register with the 2nd provider, but that's another story). Is
there a
way yo determine which direction speed-dial buttons
List,
A Cisco 7960 is registered to servers A and B, where B is the backup server,
only used by the 7960 if A is unreachable. That is the behavior of these
phones.
A call comes from server B to the 7960, which is successful. The 7960 then
tries to park the call via an attended transfer, so the
remove callprogress=yes and busydetect=yes
lotusscript wrote:
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No probs with the Snoms. No errors show on the * box
when the
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No probs with the Snoms. No errors show on the * box
when the line
asterisk-users@lists.digium.com
Sent: Tuesday, March 04, 2008 12:34 PM
Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade
Mike Hammett wrote:
I couldn't figure it out on my own. I tried to purchase a Smartnet
for the phone, but the original 7960 is not supported.
Is it technically possible
I couldn't figure it out on my own. I tried to purchase a Smartnet for the
phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have someone
remote into my network and upgrade my SCCP 7960 to the latest SIP firmware?
--
Mike
Mike Hammett wrote:
I couldn't figure it out on my own. I tried to purchase a Smartnet
for the phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have
someone remote into my network and upgrade my SCCP 7960 to the latest
SIP
, March 04, 2008 12:34 PM
Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade
Mike Hammett wrote:
I couldn't figure it out on my own. I tried to purchase a Smartnet
for the phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have
Yes it's work for me...
(with olds 7940 phones...)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Shawn
Laemmrich
Envoyé : mercredi 5 décembre 2007 23:43
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Cisco 7960 to 2 SIP servers
Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Cisco 7960 to 2 SIP servers?
Yes it's work for me...
(with olds 7940 phones...)
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Shawn
Laemmrich
Envoyé : mercredi 5 décembre 2007
Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2
different SIP servers @ the same time?
I currently have an asterisk box @ home with several sip extensions and
a Nortel C15k phoneswitch at work (not the pbx, the full phone switch).
I can connect from the SIP phone to the
On Sun, Sep 02, 2007 at 03:47:45PM +0100, Chris Bagnall wrote:
There's both a 7960 and a 7960G (and a 7961 to confuse matters further).
The 7960 is the earlier version. The easiest way to identify it from a
picture is to look at the messages/services/etc. buttons. On the 7960 the
words
Is there more than one version of the Cisco 7960?
I see some items advertised as 7960 or 7960G, but searching on 7960 only brings
up 7960G info, or ambiguous stuff.
joe a.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On 9/2/2007 at 9:32 AM, Joe Acquisto [EMAIL PROTECTED] wrote:
Is there more than one version of the Cisco 7960?
I see some items advertised as 7960 or 7960G, but searching on 7960 only
brings up 7960G info, or ambiguous stuff.
joe a.
A partial never mind, it appears they are two
There's both a 7960 and a 7960G (and a 7961 to confuse matters further).
The 7960 is the earlier version. The easiest way to identify it from a picture
is to look at the messages/services/etc. buttons. On the 7960 the words
messages and services are written on them. On the G, there's an
On 18:59, Fri 31 Aug 07, Joe Acquisto wrote:
What is involved in getting SIP firmware into a Cisco 7960 with sccp
installed?
Expensive image from Cisco? Plated in unobtanium?
You'll need the firmware and an TFTP server to get the
firmware on the phone.
--
Michiel van Baak
[EMAIL
On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 18:59, Fri 31 Aug 07, Joe Acquisto wrote:
What is involved in getting SIP firmware into a Cisco 7960 with sccp
installed?
Expensive image from Cisco? Plated in unobtanium?
You'll need the firmware and an TFTP server
On 09:17, Sat 01 Sep 07, Joe Acquisto wrote:
On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 18:59, Fri 31 Aug 07, Joe Acquisto wrote:
What is involved in getting SIP firmware into a Cisco 7960 with sccp
installed?
Expensive image from Cisco? Plated in
I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2)
If the call that I'm trying to make goes through, everything works fine. But
if
there's any sort of error (like me messing around in my extensions.conf, etc). I
can't get the connection to drop. ie: If I get the
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around in my extensions.conf,
etc). I can't get the connection to drop. ie:
Dan Austin wrote:
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around in my extensions.conf,
etc). I can't get the
On 13:52, Fri 31 Aug 07, Jason Parker wrote:
Dan Austin wrote:
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around
What is involved in getting SIP firmware into a Cisco 7960 with sccp installed?
Expensive image from Cisco? Plated in unobtanium?
joe a.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Jason wrote:
Dan Austin wrote:
Shawn wrote:
I'm having a wierd problem with a Cisco 7960 (sccp2)
and asterisk (1.4.2)
If the call that I'm trying to make goes through,
everything works fine. But if there's any sort of
error (like me messing around in my extensions.conf,
etc). I can't
Hi
I have cisco 7960 connected to asterisk ,using tftp xml config file,my
problem is it can receive any call but it cant call any extension.
Please can you send me ,how to solve this issue
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal
can you give a bit more info? I know that you need nat=never for example
- Original Message -
From: Khaled
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Sent: Tuesday, February 27, 2007 10:03 AM
Subject: [asterisk-users] Cisco 7960
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sent: Tuesday, February 27, 2007 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960
can you give a bit more info? I know that you need nat=never for example
- Original Message -
From: Khaled mailto:[EMAIL PROTECTED
---
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wireless
Sent: Tuesday, February 27, 2007 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960
can you give a bit more
Softphone Eyebeam v 1.5.2
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A.
Gombolaty
Sent: Tuesday, February 27, 2007 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960
Dear Khaled,
What
Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Cisco 7960
Dear Khaled,
What is the softphone u r using?
Thx
MAG
Khaled wrote:
I am using firmware version pos3-07-500
Kindly can you provide me with the basic configuration for cisco ip phone
and asterisk
Sent: Tuesday, February 27, 2007 2:03 PM
To:Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7960
Dear Khaled,
What is the softphone u r using?
Thx
MAG
Khaled wrote:
I am using firmware version pos3-07-500
Kindly can you provide me
I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP
Timeout Errors.
The SIP configuration files load fine.
Any ideas?
On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote:
I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP
On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote:
I've looked around and couldn't find much on this, but using two different
TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT
and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP
Timeout
On Wed, 2005-08-24 at 12:44 -0400, Asterisk User Group wrote:
I have three questions about my 7960 phone that I can't discern from the
docs/wiki.
1st - If I change the SIPxx.cnf file to change registrations it sets
up new lines as expected. If I delete a line it doesn't get removed
Cisco 7960 has six buttons/lines. Can some of them be configured for fast
dialing?
If it can't be configured on the phone, how can I configure it on Asterisk?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
Hello,
Whenever I put in a password/Shared Secret in my 7960 and try and get
it to register with asterisk on OS X setup, the phone fails to register.
Oct 31 20:03:46 NOTICE[989]: chan_sip.c:11045
handle_request_register: Registration from 'sip:[EMAIL PROTECTED]'
failed for
Which asterisk release are you running chan_skinny under?
- Original Message -
From: Will Roy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central
Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone
Before I got
I am running 1.4.0-beta2
Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST)From: Anthony LaMantia [EMAIL PROTECTED]Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
Message-ID: [EMAIL
Before I got down the path of converting a Cisco 7960 I haveover to SIP I wanted to try and set it up using Skinny.
Thephone registersok withAsterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call.
When I
Well, I've never actually been able to make chan_skinny work with 79xx
phones.
I found the chan_sccp to work quite well:
http://chan-sccp.berlios.de/
plus this patch for a problem on MeetMe (I don't remeber where I found
it, but it works!):
diff -uNr chan_sccp-20060408.org/sccp_pbx.c
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
Heh, well, I actually just started a blog to keep track of various
goings on, but I just started it so it's kinda scarce.
I intend to update it in and out with various information I email to
people so everyone can benefit from the questions and answers people
use. I'd like to see other people
Hello,
When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to
re-register themselves with asterisk, even though I put
timer_register_expires: 60 in SIPDefault.cnf
Is there a way to have these phones register themselves every 60
seconds?
Alternatively, can asterisk be made
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained). If the server goes out, they re-register after
the timeout without
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
Sent: Sunday, September 24, 2006 5:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cisco 7960 Double Natted
Hi All
Yes I know double Nat is a problem
But I have a Cisco 7960 which is remote from the * PBX ad connected via
the Internet. Each side has NAT
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