[Asterisk-Users] Cisco 7970
Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7970
Hi Greg,
Would you mind a telephone call to help me with the final steps?
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Monday, November 07, 2005 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7970
The 7970 when reset to factory will delete the firmware load leaving
just the bootloader.
1. Hold down the # key
2. Power it on
3. Keep holding the power key until the line keys blink orange down the
tree
4. Have the firmware files on your tftpserver when it boots
5. Put the load into the config file like so:
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
It will retrieve the firmware and boot.
-Greg
On Mon, 2005-11-07 at 09:50 -0500, Dan Levine wrote:
Hello
I have a Cisco 7970 phone that when I was trying to reset it to
factory
defaults it rebooted and now is stuck in a constant loop of the lights
flashing by going down the line pool one light at a time in a constant
rotation.
I have the firmware for the phone, but have no idea on how to load or
it
how to get this phone functioning again.
I would definitely be willing to pay someone to help me get this thing
back online, if someone can contact me either here or offlist to get
this resolved I would appreciate it tremendously.
Thanks
Dan
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
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[Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?
Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
I would be willing to Pay $500 for a good Asterisk / Exchange Integration Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists R 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, June 10, 2005 12:53 AM To: 'George Pajari '; 'Asterisk Users Mailing List - Non-Commercial Discussion ' Subject: RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization (b) Anyone interested if we post a bounty? Post it and I'll see what can be done. I've been thinking about this and a watcher on the Exchange server, as Race suggests, is probably the way to go. As to deleting the voicemail, probably scp or something like that would work fine. I have good experience with MAPI and CDO; I've coded an Outlook Web Access replacement for my company that works fine. Make sure you are specific in the requirements, as there are probably a couple of dozen ways this can be implemented. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?
Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Timeout
Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. Thank you for your help - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Timeout
The Box Itself doesn't get a new IP address, the router does. What I'm looking to do is have the IAX connection re-register every hour or so. Is this the right idea? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Monday, May 02, 2005 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] IAX Timeout On Monday 02 May 2005 16:07, Dan Levine wrote: Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. http://www.voip-info.org/wiki-asterisk+manager+events Connect to the Manager interface as part of the PPP script executed when you get a new IP address, and then issue an Event: Reload gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Please find me a IAX provider
Voicepulse is great... - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Monday, May 02, 2005 11:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Please find me a IAX provider Hi all, I am in Saudi Arabia, I have an Asterisk PBX with 15 PCs with softphones. I don't need incoming calls (no need DIDs). Could someone tell me who is the best IAX service provider for me? I want unlimited monthly basis or yearly basis service. my DSL is 128kbps. Thank You Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD Teliax)
Are you sure it's registering? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Gray, Jr. Sent: Saturday, April 30, 2005 12:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can't get incoming calls with IAX trunks (FWD Teliax) I have [EMAIL PROTECTED] 0.9 running, and everything seems to work well EXCEPT incoming calls. I have an FWD and Teliax trunk (both using IAX), and a Cisco 7960 SIP phone connected to Asterisk. Everything tests fine: - Can call from softphone to Cisco and vice versa - Asterisk inbound simulation works like a champ - Voicemail works fine - Outbound calls to both trunks works fine However, when I call into my system on the FWD or Teliax trunks, nothing happens. Nothing appears on the asterisk console so I'm not even sure where to start. I'm suspecting network problems, but don't know what to look for. My asterisk box sits on my LAN, behind an IPCop-based NAT router. I've forwarded port 4569 UDP and TCP to the asterisk box, but still no joy. I've googled and checked voip-info, but everything that mentions NAT as a potential problem points to IAX as the solution. Trunk-wise, I'm pure IAX (only SIP is the 7960, and it's on the same network as the asterisk box). I'm pretty new to asterisk, so if you can dumb down any debugging advice I'd appreciate it. Thanks a ton! Pat ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco SIP Firmware Price Increase
I've heard a few times that the firmware for Cisco Phones to use them with SIP is going to increase $150. Is this true? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QUICK QUESTION
Hey Everyone, How can I have asterisk ignore incoming rings so it doesn't answer a specific line. I tried setting up an empty context section but that didn't work. Thanks Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using * for Internet call waiting
I've seen this service done with AOL, I was curious how it was done on standard phone lines. Was it something the coordinated with the telco in some sort of hunt group configuration or something of that nature? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr Sent: Thursday, April 21, 2005 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using * for Internet call waiting Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. Thanks, Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 99% CPU - CVS 03.28.05
I've heard this problem could be caused by the hold music. I forgot the name of the process mpeg or wavmpeg, something along those lines... - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moody Sent: Monday, April 18, 2005 9:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 99% CPU - CVS 03.28.05 Hey Everyone, I've been running a version of the CVS without issue until late last week when suddenly Asterisk would randomly hit 99% CPU and stop registering my DIDs. If I stop Asterisk with a 'stop now' and restart Asterisk all is well... for a bit. So far I have deducted the following. Happens randomly during day and night - not at present times nor frequency Happens when no calls are present (it is a very low usage test box) If console is left open with high verbosity no errors are reported, CPU usage just climbs to 99% and the DIDs die - I only know because of Nagios and the DIDs ring busy. 'top' clearly lists Asterisk as the CPU hog. both 'uptime' and 'top' confirm the usage and the culprit. The server is at a data centre and is hardly used. It is only used for Asterisk. I have looked at all the other logs and cannot find any thing else creating entries - mail, messages, boot, anything. As I said the server does very little so it would be easy to see other entries. The Asterisk logs show nothing out of the ordinary. The machine does not have any digium hardware in it, it uses SIP for inbound and IAX for outbound. Basic calling card and voicemail functions. I can move to a newer CVS but that seems like new variables... I know this one was working and still works on a local test box using the providers. I am mainly looking to find the best way to see where Asterisk is getting stuck (some type of loop?) J ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAL FROM CONSOLE
I forgot the command to have asterisk dial and hangup from the console. Thanks everyone - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIAL FROM CONSOLE
It says no such command Dial or Hangup - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, April 18, 2005 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DIAL FROM CONSOLE Dan Levine wrote: I forgot the command to have asterisk dial and hangup from the console. exactly as you said! (help dial help hangup will show you the syntax) bye Ronald Thanks everyone - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INVALID Extension
I have a Cisco 7960G hooked up to a VPN connection tied into an Asterisk box The problem I have is after a certain amount of time, when you try to contact that extension a message comes up I'm sorry, that's an invalid extension. It works sometimes, and then it stops. I changed the Registration expiration time from 3600 seconds to 60 seconds, it helped but it's still not resolved. - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 7970
Does anyone know where to get the default firmware for the 7970. I purchased one, and made the mistake of resetting it to factory defaults without having the firmware file. I will pay someone to help me repair this problem also to help me get the unit to connect to *. Thank you Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 7970
Does anyone know where to get the default firmware for the 7970. I purchased one, and made the mistake of resetting it to factory defaults without having the firmware file. I will pay someone to help me repair this problem also to help me get the unit to connect to *. Thank you Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What is the good client softphone for windows?
Try xten.com - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Friday, April 15, 2005 6:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What is the good client softphone for windows? Hi List, What is the good client softphone for windows that connects to my Asterisk PBX which is in the same LAN? and how do I connect clients to my PBX? Thanks Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do I make Extention in my Asterisk PBX
You should definitely check out http://voip-info.org/tiki-index.php?page=Asterisk This site will put you on the right path and provide you with all the information you need to get started with * - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kumara Jayaweera Sent: Friday, April 15, 2005 6:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How do I make Extention in my Asterisk PBX Hi All, I have a working Asterisk PBX in my Linux box. now I have fair knowledge of asterisk. but not much. please tell me how do I make extensions in my PBX? Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Clients over Wan losing connection
Hi everyone, I have setup a couple of softphones outside of the internal network. They work perfectly except that after a little while you can no longer ring their extensions from the internal system to them. Asterisk just says Goodbye. I'm almost 95% sure it's because of a Nat issue. I have tried almost every type of network connection option in Xten, what seems to work perfectly is putting a VPN client on the computer and connecting to the VPN. Is a VPN my only solution? Thanks a lot Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO 7970 COLOR FROZEN
Title: CISCO 7970 COLOR FROZEN Hey Everyone, I bought a Cisco 7970 Color IP phone. I wanted to reset it back to factory defaults. I went through the sequence of holding down the pound key when the unit is powering on and then when the sequence changes to press 123456789*0#. The phone seemed to do something different after that. Now it is stuck in the constant cycle of going down the line buttons in a row of green lights. Can anyone help me with this? Thanks a million Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 Color
Hey Everyone, I bought a Cisco 7970 Color IP phone. I wanted to reset it back to factory defaults. I went through the sequence of holding down the pound key when the unit is powereing on and then when the sequence changes to press 123456789*0#. The phone seemed to do something different after that. Now it is stuck in the constant cycle of going down the line buttons in a row of green lights. Can anyone help me with this? Thanks a million Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls
Yup that works on our end as well We assign 3 of the same lines to the same phone and it works perfectly. - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Bush Sent: Wednesday, March 23, 2005 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls I get the results you want by assigning the same extension to multiple lines on the phone. Ben Friend, George E. wrote: I'm pretty new at this stuff, but I believe you will need to configure two different extensions and then roll from one to the other. Without that, it behaves like call-waiting on one line. George -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Philp Sent: Tuesday, March 22, 2005 11:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls We've just started testing with Asterisk (CVS HEAD) and a pair of Cisco 7940G phones running the SIP 6.3 firmware. One issue that we've run into is the ability to have multiple calls ring to the phone. Our scenario is that the user is using an extension and another call comes in for that extension. We'd like to have that second call ring the second line -- the same extension is configured on both lines. Currently, the second call is being sent to voicemail with the message that the extension is busy. Is it possible to configure Asterisk and the 7940 to carry out the process we'd prefer? Could someone share a dialplan that would do this? Thanks, -- Steve Philp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 79XX Phones
Title: Cisco 79XX Phones Hello Everyone, I'm trying to add a Park Button to Cisco 7960G phones connected to an Asterisk box. Does anyone know if this is possible? I'm concerned that our customer will not understand that they have to dial #700 to park a call. Are there programmable softkeys in the Cisco SIP software? Thanks so much Dan - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
