Displaying 20 results from an estimated 4000 matches similar to: "re: Polycom IP501 with Asterisk - distinctive ring"
2006 Feb 09
1
Polycom IP501 with Asterisk - distinctive ring?
This is my first foray into SIP telephony, so be gentle. :-)
The Polycom SoundPoint IP 501 phones have been fantastic so far. I still have
a lot to learn when it comes to them, but the manual seems pretty extensive
and so far Asterisk has been playing well with them.
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
2006 Feb 09
1
Re: Polycom IP501 with Asterisk - distinctive
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
2006 Feb 09
0
SOLVED: Re: Polycom IP501 with Asterisk -distinctive ring?
BOFH told me he uses it to listen to his co-workers
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith
> Sent: Thursday, February 09, 2006 12:27 PM
> To: asterisk-users@lists.digium.com
> Subject: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk -
> distinctive
2006 Feb 09
0
SOLVED: Re: Polycom IP501 with Asterisk -distinctive ring?
This feature also works on the IP301 phones. The obvious caveat is that
it is one-way only. Still nice for an "all-page" though.
-Jonathan
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith
> Sent: Thursday, February 09, 2006 12:27 PM
> To:
2006 Feb 10
4
More Polycom IP501 questions
I am starting to get the hang of this, I think. These are more
implementation questions; "is this a proper/good way of using/doing this"
kind of questions.
The IP501 has three line appearances. I have learned that shared line
appearances cannot place calls, only receive them. They're indicated by the
"half telephone" icon beside the button. Private line
2003 Nov 07
0
RE: Asterisk-Users digest, Vol 1 #1808 - 13 msgs archives gsm of asterisk ???
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
xxxx=> xxx,x,playback(file)
Ing Javier Rios
Ing de Proyectos
04167285748
212 2637246 /2637187
-----Original Message-----
From:
2006 May 01
0
Asterisk-Users Digest, Vol 22, Issue 1
Hey,
Thanks for the input Andrew. I did all you suggested but noticed that
when I did the loopback test, the output *was not* there as you
mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!").
In fact, the same message as before kept repeating every second or so:
>> Unnumbered frame:
>> SAPI: 00 C/R: 0 EA: 0
>> TEI: 000 EA: 1
2005 Feb 08
4
In-band disconnect problem (legacy PBX) - as terisk doesn't hear t he touchtone?
> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Is the channel physically being hung up before the * tone is heard?
Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't
support Kewlstart-style disconnect notification.
The sequence I hear on the extension, when I plug in an analog phone, is the
click of the
2003 Dec 08
1
Re: Asterisk-Users digest, Vol 1 #2120 - 14 msgs
In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger:
Andrew,
I modified the exten line in extensions.conf as you suggested.
Unfortunately,
It still does not work...
Ernest,
I spent approx. 4 hours reading list archives (and anything else Google
served up) on
how to configure iax.conf and extensions.conf to work with Voicepulse.
Then, I sent
an email to voicepulse
2003 Nov 25
1
Picking a channel (FXO port or SIP) for outb ound calls
> -----Original Message-----
> From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com]
> Sent: Tuesday, 25 November, 2003 08:56
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
>
>
> > Yep, we use it for international calling. Works great:
> > exten =>
2005 May 08
1
RE: Asterisk at home with Broadvoice?
RE Message: 5
Date: Sat, 7 May 2005 23:18:46 -0400
From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com
Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice?
To: asterisk-users@lists.digium.com
On May 7, 2005 11:04 pm, John Stegenga wrote:
> Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive
> ring - for a reasonable fee...
Please do a google search for
2006 Nov 10
2
Dialing from "Placed Calls" on Polycom IP501 doesn't always work
Greetings,
Has anyone noticed that attempting to place a call from the "Placed
Calls" list on a Polycom IP501 by pressing the 'Dial' softkey sometimes
simply returns the phone to the idle screen? It is not related to the
number being dialed, as we have observed two entries for the same
number, one of which worked and the other didn't.
We've experimented with calls
2005 Aug 15
1
Maximum remote directory size in Polycom IP501
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
2006 Mar 30
1
OT: Polycom IP501 and Speed Dials
Hi gang,
I know this is off-topic for Asterisk, but I don't know where else to
ask: I've setup a central directory.xml file for my Polycom IP501 phones
with a list of all the internal extensions. None of them have <sd>1</sd>
as I don't want to enable any speed dials, just have a list in each phone.
However, when a phone boots, it seems to pick a random entry and put it
2005 Oct 18
4
Polycom IP501 and record on demand
I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk
2006 Feb 08
3
Remapping Polycom IP501 buttons
Hi,
Just started using an asterisk-based PBX with Polycom IP501 phones. Am
Fairly satisfied and am starting to get into FTP setup of the phones.
Have figured out most things except for how button remapping works.
In sip.cfg, I have this entry:
<keys key.IP_500.31.function.prim="DoNotDisturb"></keys>
This works as expected but if I try to change the remapping to any
2006 Feb 08
1
Polycom IP501 MWI goes off periodically
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
2005 Jun 10
1
ATTN: Keith - Seriously OT
On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith
[SMTP:akohlsmith-asterisk@benshaw.com] wrote:
> On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote:
> > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com
> > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT
> >
> > Your MTA claimed it was called "SOURCE" but rDNS tells the recipient
2003 Oct 13
0
Call Parking and Paid Digium software modifi cations
That is how many old PBX phone systems work and it is that way our users are
used to working with the phone system. Another issue with the way Asterisk
callparking currently works is that there is only one call-park orbit, you
cannot use a different set of numbers for a different call park
instance(i.e. 700 goes to 701-720 AND 740 goes to 741-750).
We also have several Grandstream phones which
2004 Sep 08
0
Driving MWI on Norstars (was Maximum tollera ble lag/jitter...)
At the moment we're not - the email notification from Comedian Mail has
been mostly sufficient. I do however have some Dialogic D/42-NS PBX
emulation cards and the plan is to use them to set and unset the MWI lamps
based on events pushed out of Asterisk.
They may be obsolete hardware but they came in real handy for extracting the
voicemail from the old StarTalk NAM too.
Take a look at the