> Hi David,
>
> It may be IAX2 bug, do you use IAX? In my case downgrading
> back to 1.4.19
> did the job.
No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I
hear any other suggestions as to how to troubleshoot this I'll go back to that
version as well.
Best regards,
Ciao Simon,
> We are using 2 x AVM Fritz BRI cards with mISDN. The phones are
> Linksystem SPA922's and we are getting a little echo on the lines..
> from what i unserstand, these are passive cards and do not have any
> onboard echo cancellation, but im wondering if there is anything that
> can be
David Nedved wrote:
>> Hi David,
>>
>> It may be IAX2 bug, do you use IAX? In my case downgrading
>> back to 1.4.19
>> did the job.
>>
>
> No IAX for me. I don't recall ever having this issue on 1.4.19 so unless I
> hear any other suggestions as to how to troubleshoot this I'll go back to
>
Hi,
I want to put Video on Hold (3gp or other video file type )for our Asterisk,
Is it possble to use "music on hold" configuration to set up voh ?
Is there anybody can help me ?
Best regards,
PTCHEN
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Paul Hales wrote:
> I've just had half a day of wierd SIP stuff on 1.4.21.1 - most of it
> related to realtime.
> Sip peers showing up as UNKNOWN, but if you reboot the phone the problem
> goes away. For a while...
Interestingly enough, I've had my Grandstream suffering from the same
problem si
Thanks Steve,
The reset worked, and now I can access the configuration panel.
Can you give more details on how should I handle the 30 seconds issue? How
could I manage the dial plan to answer the call?
Today it works like this:
Call from PSTN comes in the ATA, it picks up the call and hot dial
I need a low monthly rate DID in Panama.
Maybe 2-3 inbound calls a day max. 1-2 outbound calls a day only.
Will be rarely used but needs to be very good quality and needs
longevity of business (eg this number is going into print so company
needs to be around for a while).
Please email w
jiangtao wrote:
> I'm using asterisk 1.4.21 and a problem with sip reg server
>
> In SIP.CONF
>
> register => 07070480800:[EMAIL PROTECTED]
> register => 07070480801:[EMAIL PROTECTED]
> register => 07070480802:[EMAIL PROTECTED]
> register => 07070480803:[EMAIL PROTECTED]
> register => test1:[EM
Hi all,
I have a little problem with my IAX phones... When I pick up the headset
I don't get a dialtone, and whenever I dial to a SIP phone I don't get
an indication tone...
Ideas?
Thanks,
David Vazquez
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Yes,
I used a Pap2 adaptor attached door tamperproof video/speaker phone. The
model I used had alarm contacts just in case it was removed from the wall
you can instant trigger an alarm system. You preprogram the extension it
dials and it waits to here a touch tone code that NO NC contacts are
acti
We have got that for $10 USD setup and $25 USD per month
If you are interested please email me back
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Friday, July 18, 2008 9:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discus
Dean asked for it so he can decide if it's worth it to him but that
sounds like the price someone would pay for flatrate and probably not
what one would want to pay for 5 calls per day.
MARK.
Sam Tam wrote:
> We have got that for $10 USD setup and $25 USD per month
> If you are interested pleas
Hi Mark,
You are correct - way overpriced. Looking for around $5-$10 a month max.
I was over estimating the number of inbound and outbound calls as well.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MFH
Sent: Friday, 18 July 2008 11:54
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten => _X.,1,Wait(1)
exten => _X.,n,ResetCDR()
; ***
Hello all! Is there a GUI for asterisk recordings other than ARI that comes
with trixbox?. I am searching for a tool to administer call recordings.
Thanks!!
Cheers!
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
[EMAIL PROTECTED]
No problem, you know you can always email us again if you have any other
requirement in VoIP>
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Saturday, July 19, 2008 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussi
How about:
exten => _9X,n,Goto(not-parked,s,1)
Doug Lytle wrote:
> Everybody,
>
> I have a fall though context that, among other things, tests to see if
> someone it trying to pick up a non-existent parked call (Defined from 90
> to 99). I have the following:
>
> [not-in-service]
>
> exten
Doug Lytle wrote:
*snipped
exten => _X.,n,GotoIf($["${EXTEN:1}" = "9"]?not-parked,s,1)
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
Richard Lyman wrote:
> Doug Lytle wrote:
> *snipped
>
> exten => _X.,n,GotoIf($["${EXTEN:1}" = "9"]?not-parked,s,1)
>
>
Close:
exten => _X.,n,GotoIf($["${EXTEN:0:1}" = "9"]?not-parked,s,1)
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Tempo
After much checking and puzzling, I cannot get my Polycom 601 to toggle
call recording with my Asterisk 1.4.21.1.
Via FreePBX, I can set a user to always record, and the recording will
show up in /var/spool/asterisk/monitor.
But if I try to start recording by toggling in-call, no luck.
I can
Doug Lytle wrote:
> Everybody,
>
> I have a fall though context that, among other things, tests to see if
> someone it trying to pick up a non-existent parked call (Defined from 90
> to 99). I have the following:
>
> [not-in-service]
>
> exten => _X.,1,Wait(1)
> exten => _X.,n,ResetCDR()
>
>
Eric "ManxPower" Wieling wrote:
> How about:
>
> exten => _9X,n,Goto(not-parked,s,1)
>
This works quite well, thank you!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
_
A followup to my own inquiry...
pig*CLI> feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
If you have interest in participating in a newly forming Colorado Asterisk User
Group, please contact Paul Gregory at BlueModus in Denver.Monthly meetings
may begin as early as the Fall 2008 and will be in the metro Denver area.
Contact information as follows:
Paul Gregory
BlueModus
t: (30
Bill Michaelson wrote:
> A followup to my own inquiry...
>
> pig*CLI> feature show
> Builtin Feature Default Current
> --- --- ---
> Pickup*8 *8
> Blind Transfer# #
> Attended Transfer
OK, I had broken the feature.conf fileset, but I just fixed it. Now I
can confirm:
pig*CLI> feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer
Actually, I thought about it for a while. What I want is something that
will allow me to restart the message if another sound is detected.
Something like this:
exten => answermachine,1,Answer()
exten => answermachine,n,WaitForSilence(1000,2)
exten => answermachine,n,Background(message)
exten => a
Hello All,
I have a request that I have not been able to figure out as yet. I need
to be able to play a beep when a call is transfered via attended transfer.
This is exactly what is in the bug tracker at:
http://bugs.digium.com/view.php?id=3819
Has any one found a way, elegant ot otherwise, to mak
Nicholas Blasgen wrote:
> Actually, I thought about it for a while. What I want is something
> that will allow me to restart the message if another sound is
> detected. Something like this:
>
> exten => answermachine,1,Answer()
> exten => answermachine,n,WaitForSilence(1000,2)
> exten => answ
I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio
drop when the audio starts on the other end of the call. So basically I
hear the first word, miss the second word and then hear the rest fine.
I've noticed this after calling multiple locations and getting some
recording on
I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is causing
it to moan that it can't set TOS. I absolutely don't want to install it
on the clien
MFH wrote:
> I'm noticing in 1.6 Beta 9 that on outgoing calls I get a brief audio
> drop when the audio starts on the other end of the call. So basically I
> hear the first word, miss the second word and then hear the rest fine.
> I've noticed this after calling multiple locations and getting
On Friday 18 July 2008 14:21:14 Bill Michaelson wrote:
> I'm preparing for a client install of * by doing a fresh one in-house.
> Unlike my earlier installation that runs asterisk as superuser, my
> current experimental box runs without such privilege. This is causing
> it to moan that it can't se
On September 26-28 in Glendale, Arizona, a group of Asterisk developers
will be getting together for three days of hacking, coding, testing,
designing and otherwise beating on the Asterisk code base. The event
will be hosted at the Renaissance Glendale Hotel and Spa immediately
following AstriCon 2
Hello,
I just upgraded my asterisk to Asterisk 1.4.21.1 I am getting this Notice
can any one tell me what i need to see in order to fix this problem.
[Jul 18 18:27:08] NOTICE[9779]: rtp.c:1286 ast_rtp_read: Unknown RTP codec
126 received from '0.0.0.0'
[Jul 18 18:27:09] NOTICE[9780]: rtp.c:1286 a
Hi,
I have asterisk 1.4.20 bridging two SIP channels with different DTMF mode
set on both. So when one SIP end points send INFO dtmf on channel 1,
asterisk is not able to generate rfc2833 dtmf events on the channel 2
bridged to channel 1. The channel 2 dtmfmode is set to rfc2833. I also tried
us
1.4 was working fine.
I thought I would try 1.6 beta 9
from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept
the call.
[Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite:
Call from 'JJ' to extension 'jj_audio' rejected because extension not found.
I change
Jerry Geis wrote:
> 1.4 was working fine.
> I thought I would try 1.6 beta 9
>
> from my asteirsk 1.4 server to my asterisk client 1.6beta it wont
> accept the call.
>
> [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite:
> Call from 'JJ' to extension 'mediaport_audio_visual' r
If you are trying to reject an IP address to connect to asterisk, there is
no need to run iptables.
Each SIP definition in sip.conf can have:
deny=0.0.0.0/0.0.0.0
permit=192.168.135.1/255.255.255.0
just set these values and it wont accept anything from that IP.
On Mon, Jul 7, 2008 at 7:37 PM, Do
> I'm preparing for a client install of * by doing a fresh one in-house.
> Unlike my earlier installation that runs asterisk as superuser, my
> current experimental box runs without such privilege. This is causing
> it to moan that it can't set TOS. I absolutely don't want to install it
> on
Just an FYI for Digium. I received a mailing today from you guys
which was nice. The price of mailing was ~$1.60 and inside was an
inflatable beach ball.
Cool, but I tried to blow up the beach ball and the the seam where the
part opens to inflate the ball was not connected to the ball
whatsoever
Hello,
Our Polycom-501 phones are set to retreive their config for the server by a
static configuation defined at the phones (boot servers). Is there any way to
change it remotely? I found no relevant field in the internal WEB browser, nor
anything in the configuration files (sip.conf and phone1
On Sat, Jul 19, 2008 at 5:48 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> Cool, but I tried to blow up the beach ball and the the seam where the
> part opens to inflate the ball was not connected to the ball
> whatsoever, so it went right in the trash.
Steve,
It is with great sadness that I read
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