similar to: Unprovisioned 7961

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

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, 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 -

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

I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in ".cop" files,

I know this isn't a specifically asterisk question, but does anyone know how to make the phone NOT use it's old config? I'm trying to get rid of the second line registration crap and it's not working. Aaron

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.

Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking

How does asterisk process URI's that get sent to it? I am having a issue with a Cisco phone, where 99% works except the call forwarding. The phone issues a X-cisco-serviceuri-cfwdall which can be seen when running a sip debug on the peer directly. However the system tries to lookup the request as a extension, aka X-cisco-serviceuri-cfwdall-<extension> And of course can't find this