similar to: CDP (was Re: network design philosophy and practice)

Displaying 20 results from an estimated 1000 matches similar to: "CDP (was Re: network design philosophy and practice)"

2008 Nov 01
1
VoIP traffic shaping
This was so interesting I had to move it to its own thread! Is anyone using this script? How does it perform compared to the older WonderShaper script? -M- ================== Thanks Kristian I will checkout the new script and see how it goes! Jonn -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at
2009 Feb 25
2
SheevaPlug Development Kit
Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I
2007 Feb 28
1
OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
Hey everyone, I came across a situation where I needed to use CDP to advertise a voice vlan to Polycom/Cisco (and other CDP capable phones) without a Cisco switch. Sure I have Cisco switches in places but I like my Polycoms to work out of the box and it isn't always practical to purchase a Cisco switch for every location. cdp-tools homepage: http://gpl.internetconnection.net/ So I
2009 Feb 04
0
[asterisk-dev] RFC 2833 DTMF w/ Level 3 Sonus
On Wed, Feb 4, 2009 at 4:00 PM, Gregory Boehnlein <damin at nacs.net> wrote: > Hello, > Is anyone running Asterisk 1.4 w/ RFC2833 to Level3's SONUS network? > We are unable to get reliable RFC 2833 DTMF working, and have instead had to > use G711ULAW w/ INBAND DTMF to get around the issue. Looks like an issue on > the SONUS side. > > Anyone else have this
2010 Jan 08
0
Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.
Hello everyone, I'm trying to turn up a SIP trunk with a Cisco UCM (Unified Communications Manager/Call Manager). It's currently configured for 3rd party call control (3pcc). The INVITEs show up without an SDP... Neither the Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified
2010 Jan 28
1
Use of "603 Declined"
Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;
2008 Dec 22
2
Using Asterisk to measure call quality: Introducing Recqual
Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: ----
2008 Nov 10
6
changing the size of voice packets
Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm
2004 Dec 23
1
Qestion about TDM over enthernet
who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list
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
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
2004 Dec 22
2
polycom and cdp
Hi, Has anyone tried to use cdp to push the voice vlan tag to polycom phones? The document says that it is supported, but I can't make it work. Thanks, Richard
2004 Sep 27
0
Cisco 7940 -60 firmware upgrades
This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it
2009 Jul 28
3
CIsco 7960 + asterisk: hepl needed
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 phone is not registered. The phone and the asterisk are in the same local
2005 Jan 18
1
Cisco 7940 Configuration
Hello all, I recently purchased a Cisco 7940 IP phone to do some testing with (to validate a migration to asterisk for our internal PBX needs). I understand that I need to update the phone for it to support SIP, so I configured the phone with an IP address and pointed it at my tftp server. When I reboot the phone I am currently getting "TFTP File Not Found SEPDefault.cnf" in the
2008 Oct 15
1
Cisco 7960 not always receiving incoming calls
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 inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest
2004 Dec 16
0
FW: Cisco 7960 (SIP) hold problems
ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.
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 21
1
Where is the * servers IP defined for sip phones?
This I am sure is a very easy question, but I can't seem to find the answer. Here is the scenario: cisco 7940g phone has SIP 6.3 firmware applied the file SIP<mac>.cnf does not seem to have a place for it: image_version: P0S3-06-3-00 #line 1 settings line1_name: "5010" ; Line 1 ExtensionUser ID line1_displayname: "5010" ; Line 1 Display
2008 Dec 31
2
Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
Happy New Year in advance by a few ticks for the northern hemisphere. Here's the first topic and guest for 2009: In any voice path there are several potential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of