Displaying 20 results from an estimated 2000 matches similar to: "One way audio - it doesn't make sense"
2006 Feb 07
0
RE: Asterisk-Users Digest, Vol 19, Issue 47
That was exactly it! Thanks you VERY much!
Mike
----
For the sip setting in sip.conf that setsup your voip provider add:
canreinvite=no
On 2/6/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote:
>
> Hi,
>
> I've had a bit of a problem with one way audio, and it happens exactly
when
> I believe it shouldn't (and works perfectly when I would guess I could
2006 Feb 23
4
Voicemail problems
Hi,
I've asked this question in the past, but I didn't get a precise answer.
Hopefully somebody will take note of my question.
Before I forget, I am using Asterisk 1.2.4.
I've been using the Voicemail app with success (i.e. it works) except for
one single thing: the ONLY message that it ever played back to the caller is
the temporary message. If I delete the temporary message
2005 May 11
3
Grandstream GXP2000 firmware update
I just downloaded the zip file from grandstreams website to upgrade my
gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files
missing on the zip file... Anybody been able to upgrade their firmware?
My website shows this files as missing:
201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin
HTTP/1.0" 200 12737 "-" "Grandstream
2005 Jul 24
2
Why can't sip/200 call sip/202
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get:
sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored
201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored
200/200 192.168.0.3 D 255.255.255.255 5060
2006 Dec 18
1
GXP2000, Linksys RV082 Firewall / NAT, Registrations
Hello,
We have several clients with GXP2000 in their network and behind NAT. We
have one particular client that has several GXP2000 behind a Linksys RV082
VPN Firewall/Router which is doing NAT services. According to SIP packet
inspection, it detects it's a symmetric NAT.
The problem we have is that even though we have configured Asterisk AND
the GXP2000 to register every 60 minutes, they
2005 Jun 08
5
GXP2000 and hint LED's
Asterisk 1.0.7
Has anyone got the hint function working, and maybe with the GXP2000.
I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment
trying to get the LED's to light up.
On ext 690, button 1 is setup for ext 691, I did this using both methods
691, and <sip:691@192.168.69.1>
On ext 691, button 1 is setup for ext 690, I did this using both methods
690, and
2006 Nov 12
2
same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension is
the only one that rings when the
2008 Apr 04
4
Advice on best operator phone (with attendant console)
One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and call
volume. We get the following problems randomly:
1. menu stops working
2. transfer key stops working
3. Line 1 LED gets
2005 Aug 01
2
*@Home/Grandstream Call Transfer
OK, now this should be really simple, but I am a bit of a newbie so please bear
with me. I have an *@Home box setup with TDM04B and two POTS lines. On the
SIP side, I have GXP2000 phones. Most things seem to work, but the users
cannot figure out how to transfer incoming calls from one extension to
another. Now I am not sure that I have things setup correctly, but is there
something
2007 Dec 06
2
Cisco power injector with GXP2000 phones
I've tried to use a Cisco power injector to supply power over Ethernet to a
GXP2000 phone without success. Although when I plugged these phone to a PoE
capable Cisco Switch it worked without a problem!
Knowing that all these three equipments implement IEEE 802.3af protocol, why
doesn't it work with the Cisco power injector? Anyone also had this problem
before?
Thanks,
Ricardo Carvalho.
2005 Jul 20
5
Grandstream GXP2000 resetting all the time
All,
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
anyone else experienced this issue? I have the phones set for static IP
addresses and that doesnt seem to help either. Any help would be greatly
2006 Mar 27
1
FW: Re: Fw: anybody has SIP realtime working ?
Actually, I have tested this here with an Aastra 9133i and an
Asterisk@Home server, and the 9133i will re-subscribe on its own after
an Asterisk reboot, if you wait long enough. It took on the order of an
hour to do so. Of course, a phone reboot will get it done faster, if
necessary, but it _will_ eventually re-subscribe on its own.
In another thread, I've seen a response that the GXP2000
2007 Feb 09
1
Problems with GXP2000 and Asterisk => Call pickupand Voicemail
1. We just dial the extension directly and have speed dials setup for
the first 6 parked positions. We don't use *8 at all.
2. Change the config on the phones under Account to "Send DTMF via RTP
(RFC2833)"
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee
Sent: Thursday, February 08,
2005 Jul 20
6
Extension Lights Patch
Guys I just read on the wiki:
"2005-07-19 - long awaited extension lights (hint priority) and call pickup
on various phones work with newly released asterisk patch digium bugtracker
- feel free to test and report findings to the bugtracker to have this
commited to cvs."
How does this work? And will it work only on certain phones or can it work
with the gxp2000?
2006 Dec 17
5
BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten => 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten => s,1,Dial(${ARG1},10,tT)
exten => s,2,VoiceMail(u${MACRO_EXTEN}@default )
exten => s,102,VoiceMail(b${MACRO_EXTEN}@default)
[ext-local-custom]
exten => 101,hint,${polycom430}
2006 Nov 15
2
Grandstream GXP2000 -- What's the Catch?
We are doing a medium sized office in NYC with 80 phones. The customer
originally requested Polycom 601 phones. The COO also authorized us to
purchase 2 Grandstream GXP2000 phones for the mail room. We find these
phones much easier to configure and work with asterisk . They support
BLF & intercom right out of the box. They can also be centrally managed
and provisioned. They also sound great
2006 May 11
2
Paging and Auto Answer on Grandstream GXP2000
I am looking to setup paging using the auto answer feature on the
Grandstream GXP2000. I am thinking I will follow the method as described
here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I will setup the 4th account on the phone to auto answer.
Does anyone else have a method that works better? I also looked at the
allpage AGI written on Voip-Info. But it seems
2009 Jan 14
1
gxp2000 and no sound asterisk 1.6
Hi
I have a grandstream gxp-2000 and trying it on an asterisk 1.6.
When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me.
How do you configure the grandstream 2000 to work on asterisk 1.6?
Regards
/ralf
________________________________________________
Ralf
2007 Oct 14
5
AA50 Paging
Hi
I just got an AA50 from Digium and the paging command reboots
asterisk when you use it. Digium says it is a requested feature and is
of low priority. Is there any other way to page 10 Grandstream gxp2000
phones with meetme or some other command than the page command.
Thanks in advance.
Kelly
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2005 Jun 08
2
IP PHONE iareaphone x100, tested??
Hi,
I have used the Budgetone 102's extensively on Asterisk and found then
quite reliable as long as you update the firmware.
The GXP2000 is quite a mess at the moment as the current firmware does
not support 3 quarters of the advertised functions and codec support is
extremely limited. I have tested the unreleased latest firmware update
for the GXP2000 and it's an incredible difference