Displaying 20 results from an estimated 6000 matches similar to: "Polycom 500 - Dialtone while connected"
2004 Oct 07
5
Broadvoice problems
Is anyone else having problems with them? Until today everything was
working fine.
But now dtmf is not working on incoming calls.
Any ideas? I tried calling them and their voicemail is not accepting
answers.
Is there another source for DIDs in the 314 or 636 area codes?
Especially a company that supports something besides ulaw.
I am going to hate switching numbers again, my wife is
2004 Sep 27
3
Asterisk Compile error
I'm trying to compile the voicemail module with mysql support and I get this
error on the chan_zap module .
Does anyone have any idea's on this one..
chan_zap.c: In function `handle_init_event':
chan_zap.c:5668: error: `ZT_EVENT_POLARITY' undeclared (first use in this
function)
chan_zap.c:5668: error: (Each undeclared identifier is reported only once
chan_zap.c:5668: error: for
2004 Dec 16
4
Polycom SIP Phones
Could someone please direct me (via personal email) to a provider with
good prices on Polycom Soundpoint IP 500's with POE cables? I need 14
of them.
Thanks,
Adam
________________________________
Adam S. Robins
Executive Vice President & CIO
PHARMACENTRA, LLP
5901B Peachtree Dunwoody Road, Suite 380
Atlanta, GA 30328
Office: 770-395-0088 x34
Fax: 770-395-0989
Mobile:
2004 Dec 20
19
Updating Asterisk
I am attempting to update my Asterisk installation from 1.0 to the
latest stable version. When I use CVS checkout, I am receiving the
following messages on chan_sip.c:
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.510.2.25
retrieving revision 1.510.2.27
Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c
M asterisk/channels/chan_sip.c
Then, when
2005 Aug 08
3
Speex QoS
Can anyone out there please tell me what ports Speex uses? I want to
set up QoS on switches but I can't seem to find this information
anywhere.
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2005 May 12
2
Inbound ANI & DNIS format
Hello,
Being totally fed up with the lack of quality and reliability from both
VoicePulse and BroadVoice,
We are switching to a direct IP connection to Global Crossing. We've
installed a local point-to-point T1 into their CO, and they will
give/take SIP g729a directly and act as the gateway for us.
In setting up the inbound SIP service, they are asking the question, "In
what format do
2005 Jun 27
6
TDM card and voicemail volume
Hello,
I saw some conversation about this in the archives, but nothing
definitive.
If a call comes in over a CO line via the TDM400P, the Comedian Mail
recording volume is so low it's inaudible. Calls coming in via SIP or
IAX do not have this problem.
Does anyone have any information on this issue?
Thanks,
Adam
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2011 Oct 14
2
Problem with outbound dialing from remote phone
I have a real head scratcher . . .
We have several employees who work from home. All have Polycom 501's that register to our office Asterisk 1.6.x server and communicate using SIP g729a. About two weeks ago, one of these remote users starting experiencing a problem with a previously working phone:
a. She could receive inbound calls,
b. She can place outbound calls to internal extensions
c.
2005 Aug 26
12
IAX2 Softphone Quality & Network Cards
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
This week we rebuilt the entire LAN with Cisco 2950-EI switches and have
employed QoS on the switches and router. Still sounds terrible.
What we are now finding is that the network card in the PC may be the
key to the problem. A Dell Optiplex P4 2.4GHz 512MB
2005 Jun 27
2
Comedian Mail User Setup Prompts
I have a user who goes into Comedian Mail for the first time and goes
thru the initial setup, changes password, records name, etc. Problem is
that every time he calls in, it thinks that it's his first time and
keeps reprompting him. His password change is reflected in
voicemail.conf. Others do not have this problem.
Where does Asterisk maintain the "first time" flag? Any ideas
2006 Jan 18
5
SAN Devices
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
Thanks,
Adam
The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent
2003 Oct 27
14
Answering Machine Detection
Does anyone have any recommendations on implementing Answering Machine detection for call generation programs?
What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example:
If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,
2006 Oct 30
2
light web user interface
Does anyone know of a really lightweight web interface that allows users to
log in and modify attributes of their extension only?
Thanks
Curt
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2006 Mar 29
1
Inter-Asterisk Using SIP
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB)
On the RECEIVING Server in SIP.CONF:
[OB]
type=user
2007 May 26
4
Asterisk in Xen domu with tdm400 hardware
Hi all !!!
I would like to install asterisk in Xen domU using TDM400 hardware.
Somebody know a howto or tutorial about that ?
Thanks in advance
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
http://www.contenidosonline.com.ar
2006 Feb 20
9
Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I was using G729 with Asterisk 1.07. With the new ability to do packet
loss correction with ILBC, I felt I'd give it a try. The new PLC does
not work with G729. I don't use Speex because my softphone does not
support it.
This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569
(IAX2). I've never really stressed the bandwidth. Typically, only
10-20 concurrent calls.
2005 Feb 10
1
No dialtone in a E1
Hi, I'm having a little problem when trying to make a call from asterisk. I
connect a SIP phone to asterisk, and in the asterisk box I have a TE110P
card connected to a E1. When a SIP client makes a call through the E1, I
received no dialtone in the SIP client.
In the same manner, when somebody from the POTS network makes a call to a
SIP client (through * and the E1) he doesn't receive the
2004 Sep 08
1
Polycom SIP 1.3.1 & Reject Button
Hello,
I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is
no longer appearent on the screen when a second incoming call comes in
unless I press the hold button on the first call.
Does anyone have a work around for this to reject a call while
continuing to talk to the first party? I should also point out that I
don't want it to be on *, as the situation varies from call
2007 Oct 03
4
Secondary Dialtone and selecting a specific line from Zap/g
I need to select a line from the Zap group channel
using the SIP Phone (not FXO and not FXS ports).
ignorepat does not work?
Also, what is the method to let the second dial tone
has another tone frequency?
Regards
Bilal
----------------
No, ignorepat is for FXS ports (FXS ports use FXO
signaling). Also,
ignorepat does not apply to SIP phones, because SIP
phones provide
their
own dialtone,
2006 Mar 22
3
Remote dialtone
Hi,
I have two asterisks connected via IAX2 trunk. The first * use dial
prefix 2XX, the second one 3XX.
Calls routing works OK.
But I don't know how to get dialtone of remote asterisk pbx.
I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of
asterisk #1 after dialing 2.
I know something about DISA but I'm not sure if it is a right way.
Can you give me advice?