Displaying 10 results from an estimated 10 matches for "tk701".
2003 Aug 25
2
Chan_h323 and a Cisco Gateway
Hi,
Can anyone tell me what should be included in h323.conf to get asterisk to
talk to a Cisco 2600 gateway? Any statement examples for extensions.conf
would also be appreciated. Thanks.
Will chan_h323 use the Cisco as a gateway anyway?
Regards,
Steven Thomas
2006 Nov 19
1
Vonage uses Cisco
...e uses for
their VoIP technologies. I stumbled across this article (although it's from
2002, I think) that suggests strongly that they use Cisco. There is no
telling what they might use in conjunction with this but this should clear
some of the conjecture.
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_case_study09186a008
00b559e.shtml
Curt
2003 Jul 10
1
Cisco 7960 SIP Craziness...
...the download, and
keeps repeatedly trying. This scenario is covered in the Cisco FAQ for
converting a CallManager 7960 to SIP. It essentially is a bug in the
firmware on the phone which requires upgrading to an intermediary, older
SIP firmware first. URL:
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a
0080094584.shtml
I don't have this, and my reseller doesn't have it handy either, though
they promise to get it (some day?).
At any rate, I've opened a Cisco tech support incident with hopes that
they'll be able to provide me the files quick and easy...
2006 May 21
1
Upgrade 7960 from SCCP 3.0 to SIP 7.5
Hi,
I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you help ?
From
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2,
I got the following:
1.
Copy the desired binary image from Cisco.com to the root directory of
the TFTP server.
2.
Specify the image in the configuration file image parameter for the
protocol to which you are converting (...
2005 Mar 26
4
Cisco's description of echo
...O to one of the SIP phones, there is no echo problem.
Sometimes when we dial from SIP --> Voip provider --> PSTN --> destination it
is okay, but other times the echo is horrible.
In trying to figure this out, I found this article at Cisco's site:
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml#1041385
It claims that echo always comes from the far end of the connection. So if I
hear echo, then the origin of the echo is in the equipment on the end of the
line near the person to whom I'm talking.
The description seems to make sense, but...
2005 Aug 11
1
PRI dropped calls w/ asterisk dropped betweenpstn & norstar
...poured over my logs most of the morning. I'm fairly convinced at
> this point the disconnect is coming from the Norstar 10 seconds after
> the call was initiated. This points to the 'T310' timer, similar to
> what is described here:
> http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_
> note09186a0080094487.shtml
> for the Cisco CallManager.
> When comparing the log from a successful call to a failed call, I
> noticed Asterisk was not passing back call-progress from the PSTN-span
> to the PBX-span.
http://bugs.digium.com/view.php?id=4468 may ha...
2006 Jun 19
0
Re: Asterisk-Users Digest, Vol 23, Issue 135
There's an excellent tutorial on Cisco's web page at
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml
It will tell you just about everything you wanted to know about echo and
more :)
The short answer to your question, however, is that echo is comprised of
two components: volume and delay. Increase either one and the problem
gets worse. In the PSTN...
2005 Mar 18
1
Cisco 7940 convert to sip
Hi!
Can anybody help me with convert Cisco 7940 CallManager Phone to
a SIP Phone? I have continious error in tftp log:
connect from 192.168.1.111
Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests
OS79XX.TXT, conversion octet
Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option
value in init packet
Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080
from=192.168.1.111
Mar
2006 Jun 19
3
ECHO Tutorial
Is there anyone that could explain to me the phenomenon of Echo or at
least point me where I can learn more? Why is this affecting the VoIP
world so much and not the regular PSTN analog world? What does the
PSTN industry have that they can handle such high volume of calls and
there is "no" echo problem?
Thanks,
Daniel
2003 Sep 24
0
Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs
...- Original Message -----
> From: "Brian West" <brian@bkw.org>
> To: <asterisk-users@lists.digium.com>
> Sent: Wednesday, September 24, 2003 2:25 PM
> Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK
>
>
>>
> http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml
>>
>> That covers the thridparty h323 stuff with *
>>
>> bkw
>>
>> On Wed, 24 Sep 2003, Sean Figgins wrote:
>>
>> >
>> > That is about what I have been seing for help. Has anyone any clue
>&...