similar to: Cisco 7960 / SIP & tftp configs

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

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

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 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

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 -

I finally got the power supply for my 7960 and am having problems getting it working. What should be in sip.conf and the SIP(macaddr).cnf file? This is what I have in SIP0002FD3BA8F7.cnf # SIP Configuration Generic File # Line 1 appearance line1_name: Asterisk Test # Line 1 Registration Authentication line1_authname: "phone1" # Line 1 Registration Password line1_password:

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

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"

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'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

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

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 have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple "sip debug" and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend

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:

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming

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 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

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

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming