search for: 7940s

...d Sent: Tuesday, January 24, 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0SIP Trunk Hi guys, I want to leave messages on our unity box. I have already converted a couple 7940s to SIP, but I can't give them out to our users because I don't want to have to deal with two voicemail systems. we have licenses for all our users on unity as is. we're about to buy a bunch more 7940s, but I don't want to cause they're expensive. I'd rather buy a ch...

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

Greetings to all. I am hoping someone can help me out with a problem I am having getting my Cisco phones, 7960s and 7940s, to download the appropriate files from our TFTP server. The TFTP server is running on Fedora Core 4. The TFTP server appears to be setup properly: service tftp { socket_type = dgram protocol = udp wait = yes user...

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality, especially on IAX2 calls, silences etc. I'm wondering whether or not the mac...

Folks -- I know this isn't directly an * issue, but I need to buy 14 7940s (preferably) (or 7960s if the price is also reasonable) --- no power cubes, immediately. If anyone has a good price, contact me offline at 512-427-1324 or lenny @ rocksteady.com Thanks, Lenny

I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this wold be accomplished? -- Rock River Internet...

Hi all, I'm having sudden MWI problems. Everything else on the phone works fine though. I have three Cisco 7940s. Asterisk server is behind a firewall running NAT. (192.168.1.202/24) Phone #1 - On the same subnet 192.168.1.250. Everything works great. Phone #2 - On a different subnet, 192.168.2.0/24. Everything works fine except the MWI. It never comes on. This is over an IPSEC VPN, but it's still beh...

...oh. This would be the trickiest, but not unsolvable. For each Asterisk connect Unity user, modify their MWI extension in their Unity sunscriber profile to dial Asterisk. On Asterisk add an extension with a small AGI script to 'touch' the local VM status. > we might as well buy the 7940s ( I wanted to get a bunch of 941s ), > or ditch unity... hrm.

I am hoping that someone has had better luck with this than I have. I would like to connect Cisco 7940s and 7960s to the Netgear FS116P, and take advantage of its POE. I know that the Cisco phones use Cisco's "pre-standard" POE implementation, but as I understand it the difference is really only two pins. I bought a bunch of PowerDsine inline power adapters which supposedly change the...

...s 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, the alternative is cisco 7940s ( around $250 ) running SCCP through the CCM. at the quantities I'm talking about, $100 is significant. Does anyone have any idea how to get this done? I've tried this: exten => 123,1,Dial(SIP/sipphone,20) exten => 123,2,Dial(SIP/ccm/3040) where 3040 is our VM pilot for ccm. but...

We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of "office-style" phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk...

I am looking for a simple way to forward calls unconditionally with Asterisk. We are running an Asterisk system with 10 extensions using SIP. One of our users leaves the office regulary, when she is out, she needs to be able to forward unconditionally to her mobile or collegue. I am trying to keep it as simple as possible, we use Cisco 7940's, they have a call forward option, when she

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and then finally go to voicemail. B...

Anybody have this work? If I dial the extension from an analog connection, it works. From a Cisco 7960 it doesn't.

I'm writing a program to quickly generate SIP<Mac>.cnf files for tftp configuration of the 79XX series of Cisco phones and would ask that anyone who is interested in using this please send me working examples of your SIP<Mac/Default>.cnf, and RINGLIST.dat files. Also, please send me your ring tones and * related logos for possible inclusion. Thanks, Christian Hoffmeyer YottaDot

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of calls sounding a b...

Hi all, On my 7940 phone when I dial out I press 9, then the number. After I press the second number (IE: 9,1) the dialtone stops playing just like it should. This is normal and similar to a regular phone. On two of my 7940s the phones continue the dialtone. No matter how many numbers you dial the dialtone does not stop until you press dial. Also, on these two phones a little X appears next to both line appearances after rebooting. They go away an unkown amount of time later. Sometimes only one will go away, someti...

I've been unable to figure out how to make blind xfer transfer a call on a zap interface by using hook flash instead of a native bridge. Is this even possible to change how a blind xfer works on the 7940s? zapata.conf: threewaycalling=yes cancallforward=yes transfer=yes

...terisk for my customers. To do this, I want to setup * for my home and business. I've been following the list for a while now and narrowed down my phone choice to Cisco. I could buy more cisco ip phones if I get the lower end 7902,5,10, etc. Am I nuts to not get a SIP enabled phone like the 7940s and up? Will using an older protocol like SCCP and/or H.323 cause me aggravation when using NAT? I've currently got a bunch of analog phones with no display. I'm also getting ready to buy a digium card and modules. Any advise would be appreciated. Maybe asterisk should add a section of...

Doe anyone have Caller ID name working on incoming calls with the following setup: cisco 2611XM recieving calls on a PRI. cisco talks SIP to the asterisk box. Phones (cisco 7940s) talk SIP to asterisk. I currently have verified the cisco router is receiving caller ID name from the ILEC, and it appears to be sending it to the asterisk. But, it looks like it comes in after the call, and * doesn't know what to do with it. A sample from the * console: -- Executing...