similar to: Call drop-out on second incoming call.

Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never

For the last few months we have intermittently been experiencing some very strange registration problems with certain polycom phones. Here is some background information: I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8 servers at different locations. Each phone is on the same network (and subnet) as the server it connects to. There is no NAT or anything else strange

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I was looking for methods in R that allow assessing the number of significant principal coordinates. Unfortunatly I was not very successful. I expanded my search to the web and Current Contents, however, the information I found is very limited. Therefore, I tried to write code for doing a randomization. I would highly appriciate if somebody could comment on the following approach.

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

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

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have

I'm trying to use the randomTF function from package palaeoSig to test the significance of a MAT reconstruction with nine analogs and a WA-PLS reconstruction with four components. I'm probably missing something obvious here but how do I make sure that randomTF is testing the reconstruction based on the desired number of analogs / components? In: fitmap.wapls = WAPLS( lumapspc,

I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file

I recently updated my phones Cisco 7960 phones (3 of them) in a high volume call place, where the Secretaries use the 7960 phones to answer inbound calls, as many as 15 simultaneous calls between all three of them. Since then I have had only constant problems, mainly that after 3 calls on a phone, if they try to xfer or do any ohter things (sometimes just answer the 4th call) the phone freezes,

Hi, I've been using R for a while now but I've got a problem with metaMDS (in the vegan package) that I can't quite figure out. I have a set of proportion data (from 0-1, rows sum to 1) that I apply metaMDS to using the command: nMDS.set=metaMDS(sqrt(test.set),distance="euclidean",k=3,zerodist="add",autotransform=FALSE) I am using a squared-chord distance

I have a 7960 with P0S3-08-2-00 firmware. Call waiting is enabled on the phone: ... call_waiting : 1 ... When I call from the 7960 to the 7960, I get a 486 busy here on the asterisk console. I can call from the 7960 to a soft phone no problem and vice-verse. I can also call from soft phone to soft phone with no issues either. Anyone know offhand what's going on? bug in the

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.class="10"/> I set up a test extension: exten =>

ivo and others, i am having a very interesting exchange about the LTP with diana. I took it offline from speex-dev as it would not interest everyone here. I will reply to ivo separately. I have also setup a mailing list on http://tech.groups.yahoo.com/group/ltpdiscuss/ if anybody is interested. but the key issues that the solves is that: 1. support p2p community based telephony that can punch

I'm having some difficulty setting up a Polycom 501 with two different extensions. I've set up in the phone's xml file the proper reg.1withlineKeys = 2, and reg.3 with lineKeys = 1. I have also set up the sip.conf file on the Asterisk box correctly. However, the phone is not making any sort of registration request to the Asterisk server. Is there something else that needs to be

Hi Andrew - > I have a need to be able to identify incoming calls based on some factor > (could be time of day, caller ID, dialed number, it doesn't matter.) -- > Assuming Asterisk can differentiate between the calls I want, how do I inform > the IP501? There are "only" three line appearances -- I can't simply just > ring a different appearance since there

Hi, we're trying to do something that's perhaps unorthodox, but seemingly should work; but not for me. We've got a Powerware 9315, that we are serially attaching one machine to. We have several classes of systems that we want to have shut down at different times. I've been doing my config testing by attaching the master machine to a Powerware 9120, and yanking the power