Displaying 20 results from an estimated 30000 matches similar to: "half channel audio after upgrade to 1.4.18"
2008 Nov 21
4
upgrade from 1.2 to 1.4 and now half channel audio
Hi all,
I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.
Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.
The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear
audio full channel.
If I do polycom to external
2007 Jul 05
1
sometimes half audio on 7960
Hi,
I am getting half channel audio on cisco 7960?
Any idea why? Details below.
Jerry
This same phone has been working for MONTHS using a TDM2400p
with no issues.
Today we got a T1 installed coming into Box A with all incoming calls
going to Box B (the TDM2400P box).
The TDM2400 is no longer in use. All incoming and outgoing calls are
done with the T1 and Box A and Box B
use SIP between them
2006 Apr 05
2
SIP Asterisk Polycom Reinvite
Wondering if anyone has experienced an intermittent one way audio
(called party can not hear) problem in these conditions;
Several IP501 phones local, same subnet.
Remote asterisk
No NAT anywhere
Polycom IP501 ulaw only, canreinvite=yes
Asterisk
Call termination path is to a sonus GSX operated by the upstream
carrier, ulaw only, canreinvite=no
The idea is that if the Polycoms are
2006 Mar 05
6
Polycom 501 power over ethernet
When I bought two Polycom 501 SIP phones, I naively thought they were
Power-over-Ethernet (IEEE 802.3af) because they were "powered over
ethernet." Silly me.
Polycom must have some odd voltage or funny way of injecting the
power, because the POE switch I bought for them (Netgear F@510P)
won't power them, though if I use the Polycom-supplied AC adapter and
ethernet power
2009 Feb 24
3
Polycom Spectralink 8002 Configuration
I have a new Polycom Spectralink 8002 and am having trouble with the
configuration or the unit but I can't see what's wrong. The unit does
not seem to even attempt to register with the Asterisk proxy but I can
make calls to it. I have viewed the syslog from the device which it
will actually write to the asterisk server so I know it can be reached.
I have also run a sip debug and
2020 Aug 06
1
asterisk 13.33 and polycom
I am using asterisk 13.33.0 and POlycom phone with the latest firmware.
The polycom phone is behind a firewall, the server is in the cloud.
If the polycom has just booted - it receives a call, after some time
(couple minutes) it no longer receives a ring. I see no errors in the CLI -
looks just like the previous call as far as I can tell.
Then reboot the phone and as soon as its ready call it
2007 Jul 03
1
Configuring BLF or Asterisk presence/Hints feature
Hi all,
I am working on
asterisk 1.2.18
zaptel 1.2.17
Polycom 650
polycom 430
SIP version 2.0.3.0131 for IP 650
SIP version for IP430 2.0.3.0127
freepbx 2.2.1
I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me.
Regards
FArooq
**********************************************
1
**********************************************
in my
2008 Apr 01
2
help with no audio
I am using asterisk 1.4.18 with a polycom phone.
sip.conf has:
[532]
type=friend
username=532
secret=XXX
dtmfmode=RFC2833
host=dynamic
context=smvoice-sip
callerid=532
qualify=no
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
I call into the dialplan and try to play demo-congrats and I hear nothing.
Firewall is disabled.
Everything is on the 192.168.1.X network for this
2006 Oct 18
2
random one way audio and noise betweenSIP phoneson same LAN
Giorgio,
I'll answer in reverse order:
I've not had reports of "noise" from my users. However, when I went down to
get the s/w version from the phone that seems to be acting up the most, the
user reported that earlier they were actually on a call that was ok then
spontaneously dropped the audio. Per my instructions (based on another
similar report I read on Digium's site),
2004 Sep 16
2
FW: Polycom IP500
I'm guessing that I need more info entered into the 'message centre'
section.
What did you key in?
Paul Hales
IT Support
Adairs
-----Original Message-----
From: Jeff Pyle [mailto:jpyle490@gmail.com]
Sent: Friday, 17 September 2004 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
I have two IP 500's on my Asterisk
2005 Jul 24
2
Busy Lamp Field SIP Phone
Does anyone have a recommendation for a good SIP phone with a busy lamp
field? I need my operator to be able to see extension status for about
20 extensions and transfer via HOLD + extension button. I've got a pair
of SNOM 360s with the sidecar, but I'm very disappointed with them. The
buttons are cheap and rubbery like a Sipura 841, the handset cord is
short and cheap, the audio quality
2006 Oct 18
2
random one way audio and noise between SIP phoneson same LAN
I'm having the same "random" problem.
I have "canreinvite=no" on all extensions. I have "qualify => yes" on all
non-NAT extensions. I do have several NAT extensions, but I've not had
reports of problems from those. 95% of my extensions (all polycom 501/601)
are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches.
In all cases, the
2005 Jan 25
5
Polycom and call waiting again..
I searched and read all the relevant posts, but I still don't have a
solution to my problem..
I've got a small queue for tech support calls using AddQueueMember. The
agents are using IP300's from polycom.
In my example, only one agent is logged int.
When a call comes into the queue, asterisk sends the call to the one agent
logged in. The agent answers and is talking to the
2004 Dec 23
1
where I can find some learning book about asterisk?
Hello ,
I learn handbook-draft.but I think I don't understand asterisk.
where I can find some learning book about asterisk?
thank u.
B.R.
John.
-----????-----
???: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]??
asterisk-users-request@lists.digium.com
????: 2004?12?24? 7:51
???: asterisk-users@lists.digium.com
??: Asterisk-Users Digest, Vol 5,
2008 Nov 07
2
help with dialplan
I have a small system, server, client and 2 phones. Phones are polycom
501's.
In general all is working fine. I can call the two phones, speak etc...
I can have the server call each phone and play a wave file.
However, when trying to setup a direct dial number of 1044 that
calls another machine running asterisk - something ODD is happening.
; This is not working....
[smvoice-sip]
exten
2008 Jan 09
2
Polycom 550 IP SoundStation Fuzzy Voice Quality
I'm setting up a new Asterisk system on a Dell server and I'm getting
"fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk
server. I've checked all of my codec settings and both the Asterisk
and the Polycom agree on u-Law encoding. I'm using the latest release
of the Asterisk code (1.4.17) and other software. If I call between
phones (i.e. two
2017 Apr 26
5
** in extensions.conf
I just tried this in my extensions.conf
exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)
Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.
Is there anyway to get the ** to work? I also am using a polycom phone if
that affects things. I'm using asterisk 13.15.0
Thanks
Jerry
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2006 Mar 23
3
Polycom 501's for sale
Converted a strictly VOIP system in NYC to NEC IPK TDM system...
will have 25 Polycom 501's for sale.
Best offer, offlist only please.
R
2009 Mar 31
2
What is the one thing that polycom can do...
On the landing page of the Polycom web site there's a "We're listening" nanosurvey, asking what is the one thing Polycom can do to improve their products. The link points here:
http://polycom.zuberance.com/survey.htm
I wrote a sentence about tweaking the user interface on the IP Soundpoint series phones, so that one can escape any level of any menu with repeated pressing of the
2005 Jul 21
2
cat 5 'joiner'? (polycom 500 problem)
What gives with the stupid stupid power system on the Polycom IP500 (I
hope they changed it on the 501's).
Anyway, does anyone know if you can get some type of joiner to connect
two Cat5 cables together, and will this affect the call/signal quality
because my Ethernet cable is longer than the dumb 11ft polycom supply
(oh and their manual is the wrong about which end goes into the handset