Displaying 20 results from an estimated 5000 matches similar to: "Polycom configs?"
2005 Aug 06
1
Voicemail -- newbie question
Hi, all
I am trying to set up voicemail. I've done it to the point where I can leave
messages.
How do I retrieve them?
Actually I have few questions:
1. I want voice mail to be available at certain extension, say 100. How do I
set it up so all users can call this number and get to their respective
mailboxes.
2. How do I let users to create their own voicemail passwords from the
phone?
3.
2005 Feb 25
1
Seting up for afirst time -- can not call
Hi, all
I am setting up Asterisk for the first time and have some problems.
Setup is very simple -- Astersik box and two Polycom SP300 phones. I will
add bells and whistles as I go, at the moment things are very simple. No
TFTP servers, so phones run with their default configuration.
I set up IP addresses, netmask and gateway IPs manually on the phones.
Now, I have read of problems with
2005 Feb 14
18
Which IP phone to use in Australia
Hi, all
I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices.
I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set.
Also I need some sort of more complex handset to be used by
2005 Feb 21
8
Minimal hardware requirements
Hi, all
I am doing "prrof of concept" system. I will have two IP phones connected to Asterisk box. Box itself will have 1 PSTN conenction and one analog phone conenction. A basic minimal configuration.
At the moment I am planning to use an old PII-350 with 128M of RAM I have lying around. I can not test anything yet, as I am waiting for phones to arrive, so question is will that be
2007 Jan 23
12
How to exit from console?
Hi, all
Stupid question, but how do you exit asterisk console without stopping
the asterisk?
Tried quit and exit:
*CLI> exit
No such command 'exit' (type 'help' for help)
*CLI> quit
No such command 'quit' (type 'help' for help)
*CLI>
Any other ideas?
I started asterisk with -cvvvvg option. Same problem if use asterisk
-r to connect. Can not exit.
Any
2005 Aug 13
14
Why NAT problem
hello
i am using asterisk-1.0.9. i have a NAT problem.
without NAT registration is ok. and if user is bhind
NAT it is registring on asterisk. but SJPhone is
showing "not registered". i think asterisk is properly
sending request to UA. any comments............this
sip.conf setting was working previously
-- Registered SIP '5000' at 0.0.0.0 port 5060
expires 120
-- Saved
2005 Jul 12
12
Any suggestions for an IP phone?
Hi all,
We are in the process of selection IP Phones to work with our *new*
Asterisk PBX.
We want to buy 4 for something less than 1000$ but with a nice set of
features to work with our mail box, lines, good sound quality, full
duplex (and maybe speaker phone).
Any suggestions for something with good voice quality and not much
troubles to setup with Asterisk?
Voici quality is the most
2006 Dec 14
4
Zaptel under FC6
Hi, all
I am building a new server. Have installed FC 6 and put in TDM400 card.
Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver is found on the system.
Is there are any known problems with FC6? I did not have much trouble
running
2009 May 06
3
Polycom Dialplan Digitmaps
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
I attempted to simply reuse the existing config files for the old phone
on the new phone, but the new phone would lock up on the 4th digit when
attempted to dial out certain numbers. So, I downloaded the newest
firmware and config templates from Polycom, and attempted to migrate the
settings. Seems I'm missing something from
2005 Sep 10
2
VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 11112
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is
2005 Aug 12
3
OT: Sendmail question
Hi, all
I want voicemails to be delivered to recepients by e-mail. I have set up
voicenail, and I can see asterisk is using sendmail to send messages out.
Using Ethereal, I can see that messages are leaving my network, but
receipeint mail server never replies back. As a result, mail delivery is
timed out.
I got a book on sendmail and it looks quite complex. It will take quite a
bit of time
2005 Jul 28
4
strange dial problem with polycom 501
I am having a strange problem with polycom 501 and dailing. I've read the
archives and no one there specifically mentions this problem, so I thought I'd
ask here.
The problem is that when the user picks up the receiver or pressed new call,
sometimes they will enter a number (for example 95072091234) and in the middle
of the number the cursor might jump back one digit. So the call above,
2007 Mar 05
4
Polycom Questions
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written for
this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of missed calls? I don't mind it keeping the missed
calls list, I just don't want that running count.
Lastly, I am trying
2013 May 05
2
My new Polycom 450's can't xfer to 4-digit extension
Hi all.
I just installed bunch of IP450's and everything went well and my
customer is happy.... except that they are unable to transfer calls to
other extenstions.
They can dial them directly just fine.
However, when the user is in a call and presses the transfer soft key,
they get dial tone, and start typing the extension, say 1008. But by
the time they get 100 typed in, the phone tries
2008 Jan 25
2
SPA3000 -- PSTN to VoIP
Hi, all
I am trying to figure out how to forward incoming PSTN call on SPA3000
to VoIP extension(s).
Basically, I have converted my home to VoIP. I have normal phone
(connected to SPA3000) and couple of IP phones. All call coming from
VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I
need to do same thing for incoming PSTN calls.
I have enabled gateway function in SPA3000 and
2005 Jul 16
2
beginners question about extension context
Hi, all
I have couple of SIP phones and they are in [from-sip] context.
I also have an IAX2 phone. I have put this one in [iax-user] context.
I want to make calls between SIP and IAX2 phones. If I put them all in same
context all is fine, however when they are in different contexts they will
not call each other and I will get message (in * CLI) that particular
extension does not exist in a
2005 Jan 12
3
Polycom IP 500 Dial Issues
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading through the administration manual to try and solve this problem,
but I do not seem to be able to find the answers to my question. I figured
I would ask here and see if anyone has some suggestions.
The problem is kind of annoying. After dialing 4 digits, the phone seems
to pause and miss the 5th digit, often
2005 Feb 22
13
TFTP Server
G'Day All,
Can anyone give me some direction in setting up the TFTP server on my
RadHat ES3 box?
I did quite a bit of reading, but I think I am more unsure now than
before. I found the information nebulous. TFTP is already installed. I
am trying to determine where the root directory for the tftp services is
located so I can copy the CISCO 7960 firmware files onto it.
Thanks.... Ferg
2006 Oct 29
4
blind transfers with IP Polycom 501
I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The
only problem I'm experiencing is the following: I can't seem to get
blind transfers to work with my Polycom 501 phones Either through the
feature code or the soft keys.
Feature code blind transfers:
I set up a feature map in features.conf like this:
blindxfer => #
This works for all my
2005 Jul 26
3
Polycom digitmap question
via google, I found the reference regarding digit maps for the Polycom phones:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html
But I don't see any instruction for prepending digits to the number
dialed. Does anyone know how to prepend a digit to the number dialed (from
the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura.
i.e. Say I want to