What handset? Some such as the Planet dont work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Monday, January 24, 2005 1:10 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Music On-Hold problem
My problem is: No
Hi All
We have Grandstream 102's running ver X.18. When hanging up after
a call has been made the grandstream seems not to disconnect
the call and when you put the handset down the phone rings
only to pick it up and be on the same call. This is happening
quite often and gets very irritating.
Can
I get the same thing. Its as if the grandstream does'nt
send a hangup signal.
Someone out there must have fixed this???
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kim Lux
Sent: Tuesday, January 25, 2005 8:57 AM
To: Asterisk Users Mailing List -
Hi All
Has any one tested Ver X.22 on the grandstreams?
If so have you noticed the problem experienced
with ringback? When you hang up the GS rings
again and its the same call you put down.
Only seen this with Ver X.16 and X.18 not yet
with X.22 but I'm still not 100% convinced.
Doug
We use the 7690 and it works fine there. Has nothing to do
with SIP as Snom, ACT, 7960 ect all work that way.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Johnson
Sent: Tuesday, January 25, 2005 5:04 PM
To: Asterisk Users Mailing List -
that you are running firmware X.22 and it is not doing
the callback when you hang up ?
Where exactly did you get that firmware version ?
Thanks
On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote:
Hi All
Has any one tested Ver X.22 on the grandstreams?
If so have
Hi
Try going into vi /etc/profile insert the lines in brackets.
USER=`id -un`
LOGNAME=$USER
MAIL=/var/spool/mail/$USER
MONITOR_EXEC=/usr/bin/soxmix
VPB_TONE=BUSY,P,400,100,500(insert the following line)
Try ACT P104
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: Monday, February 07, 2005 1:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax hardphone
Hi!
Is there such a beast
Hi
What codec are you using? Best to use iLBC, 711U/A caused
the same problem with our system. What handsets are you
using? Grandstream work well with iLBC firmware ver.11.
The problem is that there are not to many phones that
work well with iLBC.
-Original Message-
From: [EMAIL
Hi
Could anyone tell me how many lines one account can take?
If I have a Switchboard extension where all calls are routed
to with one account, how many lines can that extension take
at a time?
Cisco 7960, one account, 8 lines routed to that account?
Thanks
Doug
Hi
The
problem is not normally the phones, check that you don't have busy detect on
this sometimes can cause the phone to cut out.
What
card are you using? I have had the same problem with Digium FXO cards, we
changed to Voictronix and the problem went
away.
If you are using an ISDN
The closest we have come to this solution is to use a Cisco 7960 or the
Mitel 5055 with the flash operator panel. It works just as well as having
the buttons light up. We are currently looking at developing a flash panel
for a touch pad, with drag and drop extensions. Hey its a new dawn grow with
You need to setup an account for each Line button depending on how many you
want eg. 2 accounts = 4 lines 3 accounts = 6 lines max 12 lines.
Then tell Asterisk to ring at all the accounts you have setup for that
phone. This works well, I had the same issue.
Doug [EMAIL PROTECTED]
Hi
Thorston
It
could be the ver of Asterisk or the card driver, we have not used that
particular card but have had
issues
with that and found that the driver was the problem.
Doug
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Thorsten
Unplug the LAN and PC ports and reboot the phone
then plug them back in, login and update the firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of R A
Sent: Thursday, December 02, 2004 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
There
are a few, we use the Cisco 7960. You would need to set up an Asterisk account
for every 2 lines
though, can handle a maximum of 12 lines external and 12 lines
internal.
Doug
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Brent
with Grandstream bt100
the problem is this:
i plugin the phone but it never wake up.
there is something to do
thanks
wert
--- Doug Reid - Stormcorp [EMAIL PROTECTED]
wrote:
Unplug the LAN and PC ports and reboot the phone
then plug them back in, login and update the
firmware
Busy detect does not work as well as we would like but
if you Telco provides remote hangup detection, ask them
to activate that for you and use loopdrop.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ronald
Wiplinger
Sent: Friday, December 03, 2004 5:10 AM
Hi
1.0.5.18 is more stable than X.16. I have had reg problems
but very few and have implemented X.18 on +/- 200 handsets
without any issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen R.
Besch
Sent: Thursday, December 02, 2004 10:00 PM
To:
Hi All
Does anyone know if one could use bluetooth on a cell phone
with *? Would be nice to have your cell as an office phone
combo. I heard that there is a bluetooth module for *? If so
this should be possible?
Thanks
Doug
---
NOTICE - This message contains privileged and confidential
Hi
For desk phones I would suggest Grandstream allthough they
run at 10m/s so best to seperate the networks Voice and Data.
For exec/switchboard extentions go with the Cisco 7960 or Mitel 5220
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Sean Cook
I would say use SIP is what is suggested here:) and agree!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walid Azab
Sent: Sunday, December 05, 2004 6:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk
If you unplug the LAN and PC ports and the phone still not
starting than it is most probably destroyed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ken
D'Ambrosio
Sent: Monday, December 06, 2004 11:16 PM
To: Asterisk Users Mailing List - Non-Commercial
Thats normal when it cant discover the ID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Dent
Sent: Tuesday, December 07, 2004 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Budgetone 100 Caller ID
Hi
I feel your pain! We have had the same problem with our telco lines
but found that converting to ISDN helped. If the delay on the send
and receive two pair is to big the echo canceller is not strong enough.
Try using a Voictronix card as they seem to solve the problem to a
degree but I would
Hi
I am using a Voicetronix Openline and trying to get busy detect working
for South Africa.
Asterisk definitions are:
VPB_TONE=BUSY,P,400/500,0/500
and this works although it is to sensitive and hangs up when pressing a
DTMF tone.
If I use the Voicetronix definition:
Hi
We use the Voicetronix cards in SA and have tested them
extensively against the Digium.
I found that the Digium has many more settings that you may
tweak to get the best out of it, but have found the Voicetronix
to perform allot better. The Voicetronix has an onboard DSP which
helps with the
HI
Grandstreams have a safeguard against this. What you need to do is
unplug the eth ports and reset the phone if this does not work then
yes the phone is a dudget a new one.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen R.
Besch
Sent:
Its the version of * not the phone downgrade and it will work!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Greg -
Cirelle Enterprises
Sent: Friday, December 10, 2004 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Try [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philip
Vignola Jr.
Sent: Friday, December 10, 2004 7:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Granstream phones message button
I am
Use ISDN lines!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ken
D'Ambrosio
Sent: Friday, December 10, 2004 11:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Should echo cancellation be a science or
anart?
What handsets are you using? Could be the firmware!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ian Chilton
Sent: Tuesday, December 14, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP registrations not staying registered
Hi,
I have
Hi
Try the Voicetronix Openline 4, we found these to be the best
so far!
Cheers
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dorn Hetzel
Sent: Tuesday, December 14, 2004 5:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] least sucky FXO interface?
HI
I got the same problem that only started lately. I have to do a
stop start to get the phones registered again. One site out of 12
with the same spec.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Race
Vanderdecken
Sent: Tuesday, December
What
do you have in the message button field? should be 8500
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Gerald J.
PuhlSent: Friday, December 10, 2004 6:09 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Granstream
phones message
HI
How would I get the MWI working on the Grandsreams?
Thanks
Doug (Yip another one!)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Doug Lytle
Sent: Monday, December 20, 2004 5:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi Jacky
Try using Eye Beam from X-Ten for vidio with Asterisk.
www.Xten.com
Doug Reid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacky
Sent: Tuesday, October 19, 2004 8:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP video support problem
Hi
We use the Grandstream range, the work very well with Asterisk
although the run at 10BASET so best to keep them on a separate
network. They have all the functionality and work very well, not
the best looking phone but you get what you pay for!
Doug Reid
-Original Message-
From:
Hi
Check
the ports WAN and LAN they are somtimes mixed up we use alot of
them.
Doug ReidDirectorStormcorp Network Solutions (Pty)
LtdTel: +27 11 807 1141Fax: +27 11
807 3504Mobile: +27 83 989 0008E-Mail:
[EMAIL PROTECTED]Web:
www.stormcorp.co.za---NOTICE - This message contains
Title: SIP phones
Grandstream - are pretty loud
Mitel
5055 or the conference unit are both loud
Doug ReidDirectorStormcorp Network Solutions (Pty)
LtdTel: +27 11 807 1141Fax: +27 11
807 3504Mobile: +27 83 989 0008E-Mail:
[EMAIL PROTECTED]Web:
www.stormcorp.co.za---NOTICE - This message
The
best way to do that would be installing a Asterisk box on either
side
as the
call would only use 8k IAX per call. There will be some config on the
Cisco
side.
Doug ReidDirectorStormcorp Network Solutions (Pty)
LtdTel: +27 11 807 1141Fax: +27 11
807 3504Mobile: +27 83 989 0008E-Mail:
Hi Lisa
Is ther eany more I need to know about using 3com with Asterisk?
Other than allowing ulaw.
Thanks
Doug Reid
Director
Stormcorp Network Solutions (Pty) Ltd
Tel:+27 11 807 1141
Fax:+27 11 807 3504
Mobile: +27 83 989 0008
E-Mail: [EMAIL PROTECTED]
Web:www.stormcorp.co.za
---
Hi All
Has anyone setup a 3com SIP phone with Asterisk?
I cant seem to find a way to input the user info
such as username and password for the phone to log
on to the server.
Can someone help?
Thanks
Doug Reid
Director
Stormcorp Network Solutions (Pty) Ltd
Tel:+27 11 807 1141
Fax:+27
Hi George
Are you using a Voicetronix card, if so the driver has not
yet been updated for caller ID. They are working on it. :)
Regards
Doug Reid
Director
Stormcorp Network Solutions (Pty) Ltd
Tel:+27 11 807 1141
Fax:+27 11 807 3504
Mobile: +27 83 989 0008
E-Mail: [EMAIL PROTECTED]
Web:
Hi all
We have had Asterisk drop calls every now and then, does anyone know why
this happens? It is seldom but does happen. We have plenty of memory
in the server.
Regards
Doug Reid
Director
Stormcorp Network Solutions (Pty) Ltd
Tel:+27 11 807 1141
Fax:+27 11 807 3504
Mobile: +27 83
Hi All
Can anyone help with this message?
We are using a Swissvoice with G729 on the latest CVS of Asterisk
Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid
data (4 bytes at the end)
Apr 20 15:11:34 WARNING[5123]: codec_g729.c:196 g729tolin_framein: Invalid
data (4
Hi
We had the same problem with the Invalid Data error. We solved it by turning
silence suppression off on the handset (Swissvoice). Try looking for similar
settings on your equipment.
Cheers
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Hi All
Anyone know anything about this error? We cannot call out or in. Using a
Janghanns 4 port BRI.
Apr 26 18:32:13 NOTICE[3842]: chan_zap.c:7681 pri_dchannel: PRI got event:
HDLC Abort (6) on Pri
mary D-channel of span 4
Apr 26 18:32:13 NOTICE[3842]: chan_zap.c:7681 pri_dchannel: PRI got
Hi David
I was on site with this system and saw some other error something like this:
Avoided deadlock on zap 1-1 chan_lock. maximum retries 10
This came up between the errors:
May 3 17:43:12 pbxct kernel: qozap: dropped audio card 1 cardid 0 bytes 19 z1
71 z2 36
If you call into the
Hi All
When I phone in I get a dead line and this message, can anyone help here?
I am using a Junghanns 4 port with two ISDN BRI.
Accepting voice call from '0118071141' to '5161' on channel 0/2, span 1
May 5 08:13:59 WARNING[3839]: chan_zap.c:7514 zt_pri_error: PRI: received
SETUP message for
Hi All
Does anyone know how to set up two agents on one Snom 220, acting as
agents in a queue. When incoming call go to queue if both the accounts
on the snom 220 are busy the call stays in the queue and if one is not busy
the call will ring through.
I find that even though the second account in
Hi
Has anyone tested roaming on the Hitachi WIP-5000 and if so any
pointers?
Kind Regards
Doug Reid
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Hi All
I have a Junghanns BRI 4 port installed where only the first channel
of each line is working i.e. channels 1 and 4 work but 2 and 5 don't.
Our config is the same on this box as 15 other similar installations
where all works well. the only error I see is in /var/log/messages:
Jun 28
Hi all
Correction on my last mail, I found that line 1 both channels work
but on line 2 none work.
I have 2 BRI ISDN lines coming in on port 1 and 2 (4 channels) on a
Junghanns 4 port.
The setup by the Telco on this ISDN is different than our others, they
have 2 lines (4 channels) that are all
We had the same thing until we started using Voicetronix, it seems that this
happens when calls collide i.e... incoming call with an
outgoing? We added a script that did a soft hang-up after a call was ended
and that seemed to work ok.
-Original Message-
From: [EMAIL
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