search for: ipmid

In ipmid.cfg I have: <G3INTERCOM se.rt.10.name="G3INTERCOM" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.10.ringer="7"/> In sip.cfg I have: <alertInfo voIpProt.SIP.alertInfo.1.value="G3INTERCOM" voIpProt.SIP.alertInfo.1.c...

...") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans" voIpProt.SIP.alertInfo.1.class="4"/> Ipmid.cfg <RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1"/> Asterisk Log: -- Executing SetVar("SIP/5002-6e20", "_ALERT_IN...

Has anyone got this to work? Under Idle Display Animation, the administrators guide says "For example, a company logo could be displayed".. In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed it to 1), and under the IP 500 section, I added an entry for the bitmap that I want to display: bitmap.IP_500.66.name ="arf" but from there I'm not sure where to go...what do you change to tell the phone to actually use...

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

...I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made the change in ipmid.cfg (see below), but I am still getting this RFC error. --- ipmid.cfg ---- <RTP qos.ethernet.rtp.user_priority="5"/> <RTP qos.ip.rtp.min_delay="0" qos.ip.rtp.max_throughput="0" qos.ip.rtp.max_reliability="0" qos.ip.rtp.min_co...

Hi, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example as I have tried various entries for hours now and don't seem to be getting anywhere. Any help appreciated. Kind regards Jamie Heckford Technical Consultant

(new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone?

...ect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I searched inside polycom site (http://www.polycom.com/resource_center/1,,pw-492,00.html) and I found a link to the polycom resource center (http://extranet.polycom.com/csnprod/signon.html) but I hadn't a username and a password so I had to give up this way. I searched on intern...

Hello, I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones. Every once and a while I have problems with either dropped calls between Asterisk and my provider, or invalid RTP audio streams with phones behind NAT. I have had a few Asterisk developers look into my installation and even my provider check my setup but still am having problems. They tell me that I need to

I'm trying to develop a company phone list accessible via the minibrowser feature on the phone. The pertinent section of ipmid.cfg is as follows: <microbrowser mb.proxy=""> <idleDisplay mb.idleDisplay.home="http://server/polycom/index.html" mb.idleDisplay.refresh="300"/> <main mb.main.home="http://server/polycom/index.html"/> <limits mb.limits...

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all and allow=ulaw. told asterisk to do a reload, and tried dialing out on a zap line. It was obvious from the call quality that g729 had been selected...

I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. <?xml version="1.0" standalone="yes"?> <PHONE_CONFIG> <OVERRIDES _.0x20._log.level.change.sip="0" tcpIpApp.sntp.daylightSavings.stop.date="4" tcpIpApp.sntp.daylightSavings.stop.month="11"

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

I got em. You want em? Anyone know how I can get these to the site listed on the Wiki? Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/dfee13b3/attachment.htm

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

We are in the process of testing * for company wide deployment. We are using Polycom 300 phones, the only problem that I am running into is when I call an 800 number that has an IVR I get disconnected after about 60 seconds. Here are the logs from asterisk. I am not sure if this is a problem with asterisk timing out or if it is the phone. To me this looks like asterisk is timing out.

now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the "hands free" mode? Thanks

...hich isfor speakerphone Hmmm... Hands Free might be: voice.gain.rx.digital.chassis="15" (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under <ringType> John Matthew Marlowe wrote: > now that I have finally figured out what I was doing wrong with my > polycom phone and got it to read the configuration file Im changing > some gains. > > I successfully changed the gain for the ringer... It was too low...

...gt; >Hands Free might be: > >voice.gain.rx.digital.chassis="15" (15 is my setting) > >Call waiting? You can turn it off in sip.cfg - do not disturb settings >I think. Don't know about gain for call waiting. You might try >playing with some of the variables in ipmid.cfg under > ><ringType> > >John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael...

...tkeys.. Currently when on a call, I have to hit more and then transfer.. I'd like make transfer appear on the first screen. Right now there's hold on there.. and hold is kind of redundant, since the IP300 has a "hard" hold button. I tried doing it in the <keys/> section of ipmid.cfg, but it doesn't seem to work.. anyone done this or something similar? I checked the admin guide but am confused.. I think I'm supposed to be using the key.x.y.function.prim thing, but it doesn't work for me.. Thanks in adavance.. --Sean