similar to: Cisco Message Waiting Indicator

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

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

Hello all, I haven't used cisco phones as yet - do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/f5246238/attachment.htm

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.

I have had several cases where the message waiting indicator was stuck in the on state with Cisco 7960 SIP phones. Here are the two cases: 1. Single extension that mapped to a single voice mailbox. Restarting Asterisk or getting a new voicemail then clearing it fixed the problem. 2. Three SIP extensions that mapped to a single voice mailbox. Getting a new voicemail and then clearing it

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?

Can I get the voicemail application turn on / off the MWI (message waiting indicator) on the Cisco 7960?

Hi, I'm thinking about getting a couple of Cisco 12SP+ phones to use on my Asterisk system. I have just bought a Cisco 7960, and they are great, but too expensive to buy a lot of them, so I though I might try the 12SP+ ones. I have seen in the archives that the phones work on Asterisk, but I can't see much in there about the images in use. When I got my 7960, it had the call manager

Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they don't have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

> ------- sip.conf -------------------- > > [8991] > type=friend > username=8991 > secret=**** > nat=no ; This phone may be natted > host=dynamic > canreinvite=no ; Cisco poops on reinvite sometimes > qualify=500 ; Qualify peer is no more than 200ms > away > ;defaultip=192.168.0.15 > context=demo >

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

Hi, This might not be a very popular question, but I was just wondering if anyone have ever tried to run Asterisk on a Windows computer using Microsoft Virtual Server (http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx). I am told that you can run Linux on a virtual server using this software, so in theory it should be possible. I run a windows based domain, but am also

Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that it has a message waiting? Regards, Mike -------------- next part -------------- An HTML attachment was

I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/eae682d3/attachment.htm -------------- next part -------------- Hello Whenever I receive voicemail on CISCO or SNOM or Analog Phones (Scitec), I would like to have some kind of indication in terms of beep sound or blinking voicemail indicator....Could you please tell me the way to enable

Hi, i wanted to know that if i have a message indicator SIP phone , then MWI will work in ELASTIX ?? Let me know the Details of MWI and how test it. *--* *Thanks & Regards* *upendra* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140804/125551b7/attachment.html>

Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm