similar to: Polycom/sip.conf/voicemail configurator

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

> -----Original Message----- > From: David Gomillion [mailto:dgomillion@eyecarenow.com] > Sent: Wednesday, December 20, 2006 10:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > I think you're making it far too difficult. > > What I do is something like this: > > [outgoing]

I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on

>Message: 10 >Date: Wed, 26 Jan 2005 17:53:39 -0500 (EST) >From: "Sean A. Newton" <nester-asterisk@wewt.net> >Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again.. >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: >

Hello, You can use ZapBarge as an extension in your dialplan to listen in on conversations going on in Zap channels(Zaptel device channels) As for recording you can use the Manager interface command StartMonitor to start recording of a Zap channel and StopMonitor to stop it. Zap channels are pretty much the only ones right now that you can directly monitor and record through Asterisk. If

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

If it's already been covered, please forgive the repetition. I searched Mantis, but couldn't come up with anything. We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped working on SP IP 300s and 600s. All of them. I tried splitting the friend entries in sip.conf into user and peer. I made sure the context of the voicemail box was on the end of the mailbox option in

All, Many moons ago I set up Samba 3.x with LDAP (Fedora Directory Server) back end. Frankly, I forget what example configurations I worked off of, but the GIDs and SambaSIDs for well-known "Domain" groups (Admins, Computers, Guests, Users) start around 2512. Fast forward 6 months and we have a commercial NAS (EMC) in the domain. It has worked fine for basic home directories, but

Hello; I am using vicidial which is using asterisk 1.8, mean while when the extension has voicemail, I always see the red light on the Polycom and hear the beep sound (toot toot) in period time. Also, I can see at the LCD an option to select it for accessing the voicemail ?but I am facing the following problems: 1) The red light and the beep: How I can let the Phone only have the red light

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry

Hi, I was wondering if anyone was aware of a work around/fix for the following Polycom + Asterisk (1.2.7.1) bug, if this was doable in Asterisk, or if it is just a bug in the Polycom firmware. (Appears in both the 1.4.x release and the 1.6.7 releases) The SIP message for voicemail notify on a new message reports the correct amounts (let's say 1 new, 3 old) The phone will display new 1,

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I''m using Mechanize to parse an extraordinarily malformed html page. After submitting a form like so: page = mech.submit(dform) The result I get back is truncated. I suspect that it''s because the source HTML looks like this: <html> <head> yadda yadda</head> <p>some text</p> <html> <table yadda yadda> My

Hi folks, A user here has asked if we can display the current voicemail message's envelope (date/time/caller id of message) in the LCD of the Polycom phones we use (430 & 501). I realize this is somewhat like the many caller-id-after-the-fact threads, but I figured maybe someone had solved this a different way. Has anyone been able to do this, via caller ID, messaging, the mini-browser

I think Polycom has added another feature that nobody wants. With MWI configured, and a phonexxx.cfg that has this: <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...> </msg> Under 1.3.4 and earlier, the phone would immediately