Displaying 20 results from an estimated 3000 matches similar to: "FW: Cisco 7960 (SIP) hold problems"
2005 Jul 26
2
7960 SIP Firmware Upgrade Strange Problem
Hi,
I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go
up to 7.5
However in my first attempt to go from V.5.1 to 6.0 this is hat happens:
- The phone reboots
- The phone then reads the file OS79XX.TXT from the TFP server. In the file
I added the version "P0S3-06-0-00"
- It starts upgrading firmware
- Then I get the following message: (Upgrade Failed -
2004 Oct 04
1
Cisco 7960G w/ SIP not working properly
I have Asterisk version 1.0-RC1 running on Debian Woody.
I have 1 analog phone working, 2 inbound lines working, X-Lite is working.
The problem that I am having is with Cisco 7960 with SIP version 7.2
software. I can make outbound calls and they work fine, I even get a
notice that I have voice mail on the phone and it seems to register
properly but I can seem to dial to the phone.
2004 Dec 16
3
Cisco 7960 (SIP) hold problems
Has anyone had problems with using hold on a 7960 SIP firmware? The
problem is when the 7960 puts a call on hold and you take it off hold
again, the 7960 outbound audio is delayed on the other end. Sometimes up
to a few seconds. I've tried a couple different things, making the
"other end" a diff type of trunk ie:
7960sip --> asterisk --> IAX2 --> PRI
7960sip -->
2005 Jun 16
1
Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
Gents,
I've built an Asterisk system to replace our PBX at work and have Cisco
7960 phones (SIP 7.4) running with Asterisk 1.0.7.
How to I get Asterisk to recognise the '#' being pressed during a call?
In sip.conf I have entries likle this:
[2001]
type=friend
context=local-phone
auth=md5
username=2001
secret=xyzzy
callerid=Jack Tubby <2001>
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
2005 Feb 09
6
Cisco 7960 Beating a Dead Horse
Hi all,
So I have been reading through the docs available online and the
different threads on this list, but I cannot seem to get this phone to work.
I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached),
when I configure the phone to point to my tftp server and reboot it I
get this message:
Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750]
Read request
2005 Jan 24
0
Asterisk v1.0.1 Cisco 7960 Sip v7.3
Running those versions of code, my 7960 will not register with Asterisk.
The same 7960 is authenticating against another * server on line 2 just
fine though - with the same settings in sip.conf. On the failing *
server I am just getting 401 unauthorized errors on the console. From
the phone's shell I get that t is registering, but not authenticated .1.
from show reg.
Any ideas would be
2005 Mar 25
1
Converting 7905G to SIP
I am trying to convert my 7905G to be SIP based and seem to be running
into a few hassles. Below are all the config files and logs from the
server. I have tried to follow the pdf's from cisco and some posts from
other mailing lists that google turnedup, but it seems that nothing is
working. Am I somehow missing a fundamental step in trying to upgrade
from Call Manager to SIP?
Any help is
2004 Sep 21
1
Cisco 7940/7960 and voicemailmain not able to press keys after a hold.
I have noticed a problem with the Cisco 7940/7960 phones where if
you put your voice-mail box on hold using soft keys and come back
you can no longer navigate. I am curious if anyone else can
duplicate this problem. Happens reliably for me with the 7940
phones.
I use rfc2833 for DTMF. I would think it was a Cisco bug, but
for the fact that this did not happen with older version of
2006 Feb 28
10
A room full of Cisco 7960s behind NAT
I need to set up an office full of Cisco 7960 phones behind NAT with the
server out in Colo.
The first test phone registers fine, but the second one does not register.
The first phone's registration looks like so:
/SIP/Registry/3115552368
:64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060
When the second phone tries to register, it gets back a 404 not found. Not
a
2005 Jan 26
7
Howto Setup TFTP server on Linux for Cisco 7 960
Thanks
But how about the config files (SIP...) that needs to be inside the tftp
server, where can I get a sample of that?
That's where the images for the firmwares of the ip phones come from, on
boot right?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun
Sent: Wednesday, January 26, 2005 5:47
2005 Mar 29
0
rfc2833 cisco 7960 DTMF issue
I'm having an issue sending DTMF to cisco
dialing this extension I should hear the dtmf tone
RTP playload 101 has been sent to the cisco phone, but no audio.
in the dialplan
exten => 8603,1,Answer(1)
exten => 8603,n,sipdtmfmode(rfc2833)
exten => 8603,n,SendDTMF(1|100)
exten => 8603,n,hangup()
sip.conf
dtmfmode=rfc2833
SIPDefault.conf
I did play with all possible settings for
2004 Aug 17
1
Cisco 7.2 firmware for SIP 7940/7960 release d
Typo in your OS79XX.TXT P00 ? instead of P0S !?
-----Original Message-----
From: Michael L?jtnant [mailto:ml@zyxel.dk]
Sent: 17 August 2004 13:31
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960
released
Hi Shaun,
Saw you post, and rushed to their ftp-server and downloaded it :-)
But, I can't make my phone (7940) upgrade, so maybe you
2018 Oct 22
1
OPUS at Texas Instruments C6418
Hi Jean-Marc,
thank you for that suggestion!
It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option.
Maybe it was only part of an early version of the Opus sources?
I looked for the file in versions V1.1, V1.1.1, V1.2alpha and V1.3 but did not found it.
Do you have an idea, where I can get the
2018 Oct 22
0
OPUS at Texas Instruments C6418
Hi Robert,
The file is not distributed in the official releases, but I can find it
in the git repository.
Cheers,
Jean-Marc
On 10/22/2018 03:53 AM, Robert Madinger wrote:
> Hi Jean-Marc,
>
> thank you for that suggestion!
> It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option.
2004 Dec 20
1
Automatic forwarding of Xprint server displays?
Is it possible to enable automatic forwarding of Xprint server
displays? The -X/-Y switches just forward the video Xserver display
but it seems there is no way to get the Xprint server displays
forwarded, too.
Thanks...
Felix Schulte
--
_ Felix Schulte
_|_|_ mailto:felix.schulte at gmail.com
(0 0)
ooO--(_)--Ooo
2018 Oct 19
2
OPUS at Texas Instruments C6418
Dear Opus family,
we have implemented the Opus codec at a Texas Instruments DSP C6418.
It is working fine!
Does anyone has experience with the configuration of the codec for a speed optimized implementation on that DSP?
At the moment, we use the following settings:
#define NONTHREADSAFE_PSEUDOSTACK 1
#define FIXED_POINT
2004 Aug 06
0
I-D ACTION:draft-herlein-avt-rtp-speex-00.txt (fwd)
All:
The latest draft RTP Payload Format for Speex is available via
the IETF. See below for details.
Greg
---------- Forwarded message ----------
Date: Tue, 09 Mar 2004 15:56:23 -0500
From: Internet-Drafts@ietf.org
To: IETF-Announce: ;
Subject: I-D ACTION:draft-herlein-avt-rtp-speex-00.txt
A New Internet-Draft is available from the on-line Internet-Drafts directories.
<p>
2018 Oct 19
0
OPUS at Texas Instruments C6418
Hi Robert,
There's also a TI_C6X_ASM config option, that causes the fixed_c6x.h
header to be used, but I think it hasn't been tested in years. I don't
know if it still works, but if not it's probably not too hard to fix
(patch welcome). The fixed_c6x.h file can also probably be extended to
cover more of the C6x arithmetic operators. Beyond that, you'd have to
go to
2004 Sep 13
0
[AVT] Open Speech Repository (fwd)
interesting for anyone testing out speex :)
kfish.
----- Forwarded message from Alan Clark <alan.d.clark@telchemy.com> -----
From: Alan Clark <alan.d.clark@telchemy.com>
To: avt@ietf.org
Date: Mon, 13 Sep 2004 12:57:01 -0400
X-Mailer: Microsoft Outlook IMO, Build 9.0.6604 (9.0.2911.0)
Subject: [AVT] Open Speech Repository
We've started to build a database of speech samples in