similar to: Remote mailbox in sip.conf

Hey All, I have a user whose Cisco 7960 died the other day. When you disconnect the Ethernet cable the phone boots (as far as it can w/o network connectivity) but as soon as you plug in the CAT5 it goes dead. (no lights, no sound, no display, nothing) I tried to contact Cisco but they say I have to contact the vendor. I have sent an email to the vendor (www.thevoipconnection.com) and I am

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

Hey All, I have a two server asterisk setup. One server is in the office (which is where all the local SIP phones connect) the other is co-located and connected to 2 PRI's. The two machines communicate via IAX. The idea is to reduce the "WAN" traffic to the office. If people are off site, they connect to the remote server. Everything works fine. Now, being a gluten for punishment,

ala cisco 7960 -----Original Message----- From: Matt Schulte Sent: Thursday, December 16, 2004 10:34 AM To: 'Paul A Brown' Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems Sure thing, the biggest problem I had was getting the SIP filenames working correctly for updating the firmware (blah, I love Cisco but these phones are a joke for support). This works for me! Good luck.

I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

I just had to deal with this yesterday. I called Cisco and they gave me a part number for the support contract. I looked around and it was $90... I posted back to this list and was happy when someone gave me the correct part number, which at CDW was $10... Not too bad. Although I can't believe Cisco waists time with $10 service contracts. At that point just make the damn thing free. Anyway,

Hello list, How can you obtain the default mailbox for a SIP extension (as stored in sip.conf and shown with "sip show peer <ext>")? Is there a function to extract it? Why? Some extensions have shared mailboxes and others do not and I don't want to duplicate logic, just use the extension's default mailbox as coded in sip.conf. sip.conf ---------- [100] mailbox=100

We've been testing (aim:frziegler and aim:end1r) the Voicemail2 app for a few days now, based on a CVS build from Monday, 5/12/03-23:15. Works good! Thanks Mark! We seem to have found a bug in the MWI (Message Waiting Indicator) logic. By simply creating msg0000.txt files in both structures, e.g.: for extension 4000: voicemail1: /var/spool/asterisk/vm/4000/INBOX/msg0000.txt

The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:

I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox=XXXX but still no light. Also the messages button does not work. Any ideas?

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 -

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To:

Does anyone in Asterisk land know how to turn on the message light on the back of the earpiece of a cisco 7960 when a message is waiting? Thanks! Paul Paul Mahler mail:pmahler@signate.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040130/0efacc79/attachment.htm

Situation: * New Install of Asterisk * 7960 w/ SIP 7.4 Image * 7912 w/ SIP040406A * 3 Lines Defined on the 7960 (5104,3100,2100) Questions (configs are below): * Why won't the MWI light on the Cisco? I've tried: * mailbox=2100 * mailbox=2100@default * mailbox=2100@wvlandsales-voicemail * Does anything look goofy overall? :-) Thanks, George Sip.conf [2100]

We would like to setup in house SIP phones with numbered extensions for demonstration purposes. What is the syntax to associate a extension with SIP phone? Does the Dial application have a SIP specific entry for example: Dial,SIP/SIPphone/s|15 When I call from one extension to another I get "User is on the phone". We also have Cisco7960s to test. Currently Have X-Lite setup.

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

Hi, Has anybody been successful in running the 7914 expansion unit for the Cisco 7960G IP phone? For anybody unaware of what the expansion unit does, it provides 14 additional buttons, with an LCD display. The idea, is that with an expansion unit (a 7960 can take upto 2 of these units), a user can either assign more speed-dial's, or can monitor line status/account status. So, you can