similar to: Polycom and NAT

I hope this doesn't already exist, but I couldn't find anything to help. I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones. Does anyone have any steps on how to configure these? I have softphones working just fine, but for some reason I can't find a clear step by step on provisioning the Polycoms. Any help is greatly appreciated! Mark J.

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

All, I have a Polycom 650 phone, when turned on displays "Checking application". Can any give me some information as to what is wrong? I have copied the CFG files from a 601 phone to work with this 650. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/daadda2a/attachment.htm

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment

Hi All, I have a site using Polycom 501 phones and for some reason the caller ID of the phone number is coming up as sip:<number>@<ip of server> Does anyone know why? It seems to be a Polycom thing as a Linksys phone displays the CID number as just the number. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

has anyone done it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080528/ff622d28/attachment-0001.htm

All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As

Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

Hi All, I'm not sure if this is related directly to asterisk or not but on my Polycom 320 when I try to dial a number smaller than 4 digits I get an error on the phone saying "Enter more digits". The dial plan section is listed below. <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1"> <digitmap

Hello all, As I've been exporing libvirt api, I've seen event registerers, and types of events are defined, undefined, and you know the rest. Is it all for event types? Any ideas about how to look for a device access event, for example? Kind regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

I've just wasted a day chasing my tail because of subtleties introduced to handle the optionality of the DataLayout. I would like to never do this again. =] We now have this attached to the Module with just a flimsy faked-up pass to keep APIs consistent. So, is there any problem with beginning down the path of: 1) Synthesizing a "default" boring DataLayout for all modules that

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry