similar to: Cisco 7940/7960 phone and conference calling ?

Hello, Anyone else having problems with the Cisco 7940/7960 (5.3 firmware) and the latest CVS build, placing conference calls from the phone? I've noticed the party on the Cisco phone's side will sound very garbled, and delayed by several seconds. I haven't begun troubleshooting yet, though I'm able to reproduce this easily... Thanks in advance, -a

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

> -----Original Message----- > From: Travis Johnson [mailto:tlj@ida.net] > Sent: Monday, September 08, 2003 1:05 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Cisco 7940/7960 ethernet ports > [...] > We are having a problem with Cisco 7940 and 7960 phones when > the PC is plugged into the 2nd ethernet port on the phone. It > will drop the

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

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

Those of us that use Cisco 7940/7960 sip phones know that we've been impacted by two very different changes that have occurred over the last couple of months. First, when cisco created sip v6.x code, they implemented a new DSP (as well as other software changes) that effectively drops any incoming rtp packet that does not have even timestamps within the rtp packets. The dropped packets

Has anyone had any major issues with the Cisco 7940 and or 7960 phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030818/2c156d1f/attachment.htm

Hi All, I need some help w/supervised transfer and conference w/a 7940 phone. When I do a blind transfer the calls go through great, but when I do supervised transfer the 7940 tells me "Transfer Denied". When I do a conference call I hit the "conf" key and then dial the next extension. The new call connects and I hit "conf" again but the calls do not get bridged.

Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

We've been experiencing an odd issue lately. I'm not sure when it started because it's not happening on most calls--it seems confined to a couple of our queues. It's consistent though. Here's the CLI output: -- Got SIP response 400 "Bad Request" back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions

Just in case anyone here hadn't noticed, Cisco is apparently making 7940/7960 SIP 8.2 firmware freely downloadable by anyone: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 username: anonymous password: your email address -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada

Hello list, is there anyone out there that could grab the new SIP firmware 7.5 for the 7940/7960 from Cisco's Site and mail it to me (r.zagler@fog.at)? i already ordered a support contract but did not get my access data yet! Thanks, Roland

It doesn't seem as much broken as just annoying. I am holding off on upgrading until this resolves, but it doesn't seem to affect performance, anyways. BTW, some folks say that the server address only gets appended to the CID when a redirect or something comes about. Our experience here shows that the IP always gets appended. >Alexander Burke wrote: >> Just in case anyone here

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

The documentation of how to use the 79xx series' phones and features with Asterisk is really hard to find and put together. The higher end phones like 7970 are more like converged PC+phone, a thin client to telephony and network apps. But it's really hard to target it as a development and deployment platform because the docs and techniques are so obscure. There seems to be a fair amount

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

If I relay through my Cisco 7940/7960, does it do QOS, even with a dumb switch? I know you can set quality/qos but only if you have a layer2/layer3 switch that supports the tagging. A simple little linksys 5 port switch wouldn't know about QOS, it'd give everybody equal priority. If a computer plugged into the phone, and the phone into the dumb 5 port switch and then to the internet,

Hi all, Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones with SIP. I'm using also op_panel 0.25 (snapshot). I'm using * queues. I want to properly implement DND via *78 and *79. I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd variables and this is fine for FOP. The DND works in normal cases, since I catch it with my Macro dialsip, HOWEVER

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but