similar to: 7960 backup proxy registration

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.

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 -

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

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

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.

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

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

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

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

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.......

Hello all, I've got a Cisco 7960 running version 7.4 firmware (heard there were problems with 7.5) and I can't get it to register with Asterisk. I've stripped down my configs on the phone to a bare minimum, and posted them below. Basically, the Cisco phone sends absolutely no packets to the proxy when it gets booted. If I make an outgoing call I see traffic getting to Asterisk, but

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031208/0128c586/attachment.htm -------------- next part -------------- Thanks alot. I found the problem, with SIP firmware image was 2.* large size SIPDefault.cnf leads to Buffer overflow, I kept firmware_version field and deleted rest. That helped in upgrading the firmware to higher version.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

> -----Original Message----- > From: Babak Pasdar [mailto:bpasdar@IGXGlobal.com] > > I was wondering if anyone else has had this problem. I have purchused > several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run > accross 2 that that give me malformed TFTP and refuse to upgrade to the > latest version of SIP code -- 5.3. In fact some of the other phones also

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

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described

Oh, that's great. It wasn't clear from that page, at least not for me. :-( Having it clearly stated on the document would save me (and probably others) lots of time. Thanks for clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua

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