similar to: CIsco 7960 + asterisk: hepl needed

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

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 -

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'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

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:3115552368@64.169.xx.yyy:5060 When the second phone tries to register, it gets back a 404 not found. Not a

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

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47

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

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 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"

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.

Dear All, I'n trying to make a simple call forwarding, however I have small problem when evaluating an expresion. Here is my extensions.conf ... ; Unconditional Call Forward exten => _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _#21*X.,2,Hangup() exten => #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})}) exten => #21#,2,Hangup() ... exten =>

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

Can you also post information such as: Type of phone (model Number would be idela) How is it conencted, SIP, ZAP, IAX, Channel Bank. Corresponding config files would also help. Help us help you. >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >>Paul A Brown >>Sent:

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.

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.......

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:

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

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6