Displaying 20 results from an estimated 6000 matches similar to: "What have I misconfigured?"
2005 Aug 11
4
Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone
You are right.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo,
Louie
Sent: Thursday, August 11, 2005 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone
You write out a
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:
[sipphone]
type=peer
2005 Aug 02
9
Polycom phones w/ two lines on different servers
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
2005 Aug 04
5
newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's
on an asterisk server. I am working on my
extensions.conf and am trying to make it so that all
my extensions can dial each other. My extensions are
number 720, 721, 722, 723 ..etc
in my from-sip context I began doing entries such as:
exten => 720,1,Dial(SIP/720,20)
exten => 720,2,Voicemail(u720)
exten =>
2005 Jun 15
3
Includes include the includes?
I am grouping my extensions by building like so:
1XX is Building 1
2XX is Building 2
7XX is Office
[Office] extensions has the following includes
7xx
Include => Local
Include => International
Include => Building1
Include => Building2
[Building1] has
1xx
Include => Office
Include => Building2
Include => Local
I done't want building1 to access international, but does
2005 Sep 06
5
Good Polycom Dealer?
Could any of you provide me information on a good
Polycom phone dealers to utilize. One who provides
firmwares ..etc
Thank you!
Kenny
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2006 Jun 21
4
Polycom 601 problems with multiple registrations
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
2007 Jun 07
3
Polycom phone registration problem
Hi,
One of my users is in trouble with his polycom phone hooked to an
asterisk server.
The phone works fine for a few days, and then disappears from the
registered sip peers in asterisk.
The user is able to place outbound phone calls, but can't receive
incoming calls until the network plug is unplugged/plugged.
Working line
XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060
2005 Sep 27
2
Polycom IP 500 - problem dialing extra numbers
hi there
I'm setting up asterisk@home and I'm using Polycom IP 500 phones.
When I call a number that has a digital receptionist (i.e. "dial 1 or
such and such, dial 2 for this and that...") the Polycom doesn't seem
to send the extra digits. When I try it with X-Lite things appear to
work fine, so I think the problem is with the Polycom configuration.
Here's some
2005 Sep 06
2
Polycom ip301 hangs at Running "sip.ld"
My polycom phone is now hanging at Running "sip.ld".
I modified it's config via the web interface to register with my
asterisk box.
I have tried to restore the default settings wth 468* and it doesn't
seem to work.
Any ideas?
-jonathan
2005 Aug 16
3
Can not dial more then 23 calls
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.
We can not get more then 23 total active calls to connect to the test
numbers, the test numbers
2005 Aug 11
9
Polycom IP301 and 501 with asterisk...
Hi,
I am about to buy ip pbx asterisk system but what ip phones do you
recommend? Are polycom ip all functional with the ip pbx system???
Be waiting.thanks a lot
Marlo
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2007 Jun 09
3
Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Hi all,
My company has pretty much standardized on Polycom phones and I am in
the beginning phase of writing a GUI for administering/managing polycom
provisioning at multiple sites which we intend to release as OS. I've
started studying the docs and I'm having trouble understanding the
following xml attribute:
voIpProt.SIP.requestValidation.x.request.y.event
I understand what it
2012 Feb 10
3
Polycom firmware 4.0.1 and paging
Hi,
I just moved many Polycom phones from firmware v3 to 4.0.1b. Anto-Answer
simply stopped functioning. I can downgrade and make it work, upgrading
kills it again. There obviously is a difference in how the newer firmware is
treating this auto answer sip header.
Can anybody tell me if they have Polycom firmware 4.x.x working with
auto-answer/paging? Just so I know it's worth my time
2006 Nov 07
7
should_redirect_to in advance - feels unnatural
I can understand that it''s easier for rspec to set up a mock in advance
of the controller call. But it makes it difficult to do something like:
context "The HarkController, given Louie the logged-in user" do
setup do
post :login, :username => ''louie'', :password => ''atest''
end
specify "should redirect Louie to the home
2012 Jan 05
1
Blind transfers being cancelled by asterisk & hanging up on remote caller
Hello all,
I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that
2011 Aug 08
2
Polycom and auto answer
Hi,
I've been meaning to fix my non-working paging feature here for a while, and
I've just spent the last 5 hours looking at many, many web pages that all
say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both
older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with
latest 3.3.1f).
I have changed the correct values in sip.cfg like
2004 Sep 02
5
Polycom SIP INFO & Changing Ringers
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 =>
2005 Jun 14
5
HT-488 vs. SPA-3000?
Hello,
Just want to tap the collective wisdom of this list as to experiences
pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters...
Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be
the top of the pick..Any comments and experiences esp. with Asterisk
compatibility would be great, before I plonk in the bucks.
TIA.
/wai-sun
2011 Feb 24
2
Paging with Polycom 3.3.x
Hi,
My phones stopped auto-answering when being paged, since I moved on to
Polycom firmware 3.3.0 (3.3.1 is the same, I tried). That is with Asterisk
1.6.2.16.
I looked at the wiki but nothing I try there works, even if I cut and paste
the same setup.
Any one has any idea of what I should change from my old 3.2.3 setup? My
older phone (501) still using 3.1.6 still auto-answer