similar to: Cisco 7960 SIP Upgrade

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

Hello all, I feel dumb asking this, but does anyone have a link to the SIP firmware for the 7912 on Cisco's site? I have a SmartNet contract, but I just can't find the link (you can search for "7960 sip firmware" and find that fast). Thanks for the help, Jeb Campbell jebc@c4solutions.net

I was going to setup my extension on my employee's phone so he could answer calls as well as myself. I noticed that once he registered, I could no longer receive calls on my own phone. Is this a limitation of Realtime or something else in Asterisk? I've had multiple devices register to the same definition somewhere else before in Asterisk. If I can't do it that way, I'm

Just a quick note to people looking for SIP firmware images for Cisco phones: To access these files from Cisco's website, you need to have a Service Contract (SmartNet) on at least on of your phones. I though a contract was several hundred dollars, but it is way cheaper! Their lowest level, 1 year Next Business Day 8x5 SmartNet program, costs around $70.00, and gives you unlimited access to

I don't know everything that's going on as someone else has been working on the project, but it hasn't really been going anywhere, so I had some questions. We've got some Snom 320s with Asterisk 1.2.9.1 (I believe). All was well (with a previous release), but the phones started to get real choppy. We are also running a softphone at this location and it was fine. The SIP qualify

What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware

Hi, We've just bought a new cisco 7965g and web are trying to connect it to asterisk. I've bought smartnet and downloaded cmterm-7945_7965-sccp.8-3-3SR2.exe cmterm-7945_7965-sip.8-3-3SR2.zip The zip file contains: Archive: cmterm-7945_7965-sip.8-3-3SR2.zip Length Date Time Name -------- ---- ---- ---- 2496963 11-05-07 10:06 apps45.8-3-3ES2.sbn 585536

I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT.

My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I

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I have multiple upstreams in my office. The primary upstream is having some issues with latency\jitter. I want to move the VoIP traffic to another interface. I have the router set to send all traffic destined for "local" networks out the respective interfaces. Traffic destined to the Internet goes out one of the upstreams. I can do this on a per-IP basis and have successfully done

My provider has a Coppercom switch. I have included the authentication information they gave me. How would I structure this in Asterisk to the registration and the entry in sip.conf? User Name - 8159093010 Password - XXXXX No Pin Proxy - sip.essex1.com (10.1.3.2) Outbound Proxy - proxy.essex1.com (63.164.210.14) Change setting to use "outbound Proxy" ---------- Mike Hammett

I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part

Can someone point me in a direction where I can acquire the Cisco 7960 SIP firmware? I just purchased two phones of ebay, but they are both in Call Manager mode. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/6265b777/attachment.htm

Hi, I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and run sip to an asterisk server, but have their voicemail on Unity. these phones are $150 each,

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

I thought I read somewhere that you no longer have to do a special make command for the 2.6 kernel. Is this true, or should I still make linux26? I'm having problems getting anything zaptel to load properly. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Those of us that use Cisco 7940/7960 sip phones know that we've been impacted by two very different changes that have occurred over the last couple of months. First, when cisco created sip v6.x code, they implemented a new DSP (as well as other software changes) that effectively drops any incoming rtp packet that does not have even timestamps within the rtp packets. The dropped packets

Hi, I have followed the thread on where to get a maintenance contract so that I can upgrade the firmware to a SIP version and have called two different places, CDW and Insight. Neither place will sell the contact, all I get is a B.S. excuse to why they can't. Has anyone had luck with a more friendly place or do they have the images? TIA, Trevor