similar to: Called Party Identification on Polycom IP501

I remember seeing something like this on the list a while ago, but I'm darned if I can find it. We have a number of Polycom IP501 phones, some of which have more than one registration on them. When a voicemail is left for a phone with only one registration, the MWI lights up and stays lit until the voicemail is listened to. However, on our phones with more than one registration, the MWI

Greetings, We are trying to make our corporate directory (around 400 entries) available via TFTP to some Polycom IP501 phones. A small (~40 entries or so) file works, but the full file fails to load. Does anyone know what the upper limit on directory entries is? The size of the XML file itself is only 60K - you'd think that would all fit into the phone with no problems..... I would

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Hi Guys Just a quick question regarding on the 501, has anyone been able to configure the transfer button and messaging buttons to work with asterisk? Can you share a configuration to do this? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/df504eef/attachment.htm

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

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

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

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?

Hi! I've got MWI working just fine for my 501, but it's on if I have -any- VM messages. I only want it on if there are *new* messages. Any ideas as to what I should be changing? Thanks! -Ken

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or

Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100

Hi R-helpers, I would like to subset my dataframe, keeping only those rows which satisfy the following conditions: 1) the string "dnv" is found in at least one column; 2) the value in the column previous to the one "dnv" is found in is not "0" Here's what my data look like: ??? POND_ID 2009-05-07 2009-05-15 2009-05-21 2009-05-28 2009-06-04 4 ? ? ? 101 ? ? ?