similar to: Asterisk - Different Subnet to phones (Cisco 7960)

Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put "P0S3-04-4-00" in the OS79XX.TXT file, the phone downloads this

Hi all, i have a problem with some cisco 7960. Yesterday i did a firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the most documents. Now i get the message Phone unprovisioned and in TFTP-Log i find the following line: 07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin to 192.168.0.83, 0 bytes So i think it now wants to load the file P0S3-07-3-00.bin. But i have not

Here's a clip of comments lifted from a Cisco bug list. This will be perhaps useful to those of you who have just purchased a Cisco phone off eBay. JT ------------- (1) Short problem description: Documentation on how to load SIP image on phone with skinny software (2) Longer problem description (what happens): If the phone is loaded with the Cisco Skinny code, then there is a small

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

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

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

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone.

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

Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert

Hi to all, I've just received my Cisco 7905G ipphone. I want to connect it to asterisk server but it looks that it has been preloaded with sccp protocol, so I guess I need H.323 or SIP firmware image of some kind. I have a working tftp server on my asterisk box also....What do I need to do now to get things wokring? Thanx in advance, Victor...

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Im having problems setting up a trunk between two locations. Heres the setup I have: Server A is connected to the PSTN at my datacenter Server B is connected to a clients e1 line at his datacenter I only want to route calls from Server B to Server A and out through the PSTN. Server A has a lot of other things connecting to it, so I need a very specific context for all calls to go through.

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

Please forgive me my ignorance ... I've spent two days trying to find out something about the format of the default configuration file, which CCM produces. The only example I have so far is the one from the chan_sccp source. There were tons of references on entering the callmanager commands on a cisco command line - which I don't have (don't need thanks to chan_skinny + chan_sccp).

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

I am wondering if it's possible to use a bunch of cards in a PCI backplane instead of going out to the extensions with T1 and then and adapter. How are people connecting to large amounts of extensions?