On Wed, 12 Nov 2008, Julian Lyndon-Smith wrote:
> Man, I really feel stupid, but after banging my head on a brick wall for
> several hours ... I need help!
AIUI - Pickup works on an extension..
So if the xlite is SIP/5608, but extension is 444608, then you need to
pickup 444608.
Gordon
>
> I'
Arrgh. this is driving me nuts. Can anyone put me out of my misery ?
Pretty please ;)
Julian Lyndon-Smith wrote:
> Man, I really feel stupid, but after banging my head on a brick wall for
> several hours ... I need help!
>
> I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten
mp3player, is just for your need,
use it this like
exten => _X.,1,mp3player("http://www.test.com/test.mp3";)
try this page
http://www.voip-info.org/wiki-Asterisk+cmd+MP3Player
best
--- On Wed, 11/12/08, Singer X.J. Wang <[EMAIL PROTECTED]> wrote:
From: Singer X.J. Wang <[EMAIL PROTECTED]>
Subje
On Tue, Nov 11, 2008 at 06:02:49PM -0800, Don Fanning wrote:
> Greetings,
>
> I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover
> cable. I got it where the alarms are all ok/green but I'm unable to
> dial out or dial into the AS5200.
>
> Anyone have any suggestions as to wh
On Tue, Nov 11, 2008 at 07:05:23PM -0500, Jim Duda wrote:
>
> >> When it fails, I get this message:
> >> [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of
> >> type 'DAHDI' (cause 0 - Unknown)
> >
> > Can you enable debug logging? Do you see any message about the casue for
>It wouldn't hurt for you to do a code review on them,
I'd better get more up to speed on * in general first. It would be interesting
to compare them to my code. However, I don't have a useful * installation
here, yet--I'm working on it.
Wilton
__
You'll need to lose the double quotation marks in the assignment:
Set(CALLERID(name)="Fred") becomes:
Set(CALLERID(name)=Fred)
If it still doesn't work, then it means that your particular provider
does not support the ability to be able to set the caller ID name, or
it's receiving a corrupted
sean darcy wrote:
> I've tried to create a subroutine that sets callerid name based on number.
>
> extensions.conf:
>
> ...
> exten => s,1,Answer()
> exten => s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
> exten => s,n,Dial(${mainline},60)
> ...
>
> [set-callerid-name]
> exten =>
C F wrote:
> Who you calling? Is it a remote non PSTN phone number? Or a PSTN number?
>
It's incoming. Both pstn and voip.
sean
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or up
On Tue, 11 Nov 2008 18:26:01 -0800, Paul Chambers wrote:
>Anyone have practical experience using inexpensive GAP-compliant DECT
>handsets with the Snom M3 basestation?
>
>When I asked Snom support, the answer was that 'basic functionality
>should work', but they didn't elaborate. I'm _guessing_
Anyone have practical experience using inexpensive GAP-compliant DECT
handsets with the Snom M3 basestation?
When I asked Snom support, the answer was that 'basic functionality
should work', but they didn't elaborate. I'm _guessing_ that means
registering/unregistering with the base, making cal
On 12/11/2008 6:20 a.m., Tilghman Lesher wrote:
> On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
>> On 11/11/2008 10:48 p.m., samuel wrote:
>>> So far I've updated a few machines (1.4.22) and the DNS queries are
>>> reduced to a minimum, at least haven't seen DNS channel queries...
>> Sli
Greetings,
I have a AS5200 that I have interfaced to a T100P via a T-1 Crossover
cable. I got it where the alarms are all ok/green but I'm unable to
dial out or dial into the AS5200.
Anyone have any suggestions as to where to begin troubleshooting this?
___
A little more information:
If I change the dial command to
..snip..
exten = > 444608,1,Set(__PICKUPMARK=5608)
exten = > 444608,n,Dial(Sip/5608)
..snip..
and the pickup command to
exten => _**,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten => _**,n,Hangup()
then it works ...
Julian
Julian
Instead, they are likely releasing something newer and better. I believe
they have always had SIP software for download, however, it is never the
most recent. They only provide 'previous software' for end-users, if you
want the latest, you still have to go to your vendor.
http://www.polycom.com/us
Man, I really feel stupid, but after banging my head on a brick wall for
several hours ... I need help!
I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten
5707, and I've got an xlite on 5608.
When I make a call from an outside line, I dial SIP/5608. The little
blinky light o
Wilton Helm wrote:
> I'm a bit puzzled, also, having implemented ulaw and alaw in an
> embedded application. Each can be done with a 16 Kbyte table in about
> 0 time with no errors. There are probably tricks that will cut the
> table down by 2 or 4 X for a small cost in CPU cycles. The invers
>> When it fails, I get this message:
>> [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of
>> type 'DAHDI' (cause 0 - Unknown)
>
> Can you enable debug logging? Do you see any message about the casue for
> that?
Yes, I enabled logging, however, no additional logging was a
It apparently isn't built with IMAP support. That would be a bug in my
packaging. I'll see what I can do with it.
Jason Lixfeld wrote:
> I'm having some issues getting app_voicemail_imapstorage to talk to my
> IMAP server. From imapstorage.txt, I've got the voicemail.conf
> configured prope
On Tue, 2008-11-11 at 16:11 -0700, Wilton Helm wrote:
> I'm a bit puzzled, also, having implemented ulaw and alaw in an
> embedded application. Each can be done with a 16 Kbyte table in about
> 0 time with no errors. There are probably tricks that will cut the
> table down by 2 or 4 X for a small
I'm a bit puzzled, also, having implemented ulaw and alaw in an embedded
application. Each can be done with a 16 Kbyte table in about 0 time with no
errors. There are probably tricks that will cut the table down by 2 or 4 X for
a small cost in CPU cycles. The inverse requires 256 16 bit words
I'm having some issues getting app_voicemail_imapstorage to talk to my
IMAP server. From imapstorage.txt, I've got the voicemail.conf
configured properly, but if I leave a voicemail for extension , I
see no indication that the module is trying to reach the IMAP server.
What am I missi
There are new release candidates for dahdi-linux (2.1.0-rc3), dahdi-tools
(2.1.0-rc3), and dahdi-linux-complete (2.1.0-rc3+2.1.0-rc3, a combination of
dahdi-linux and dahdi-tools in one package) that contain a new DAHDI driver
for the B410P Quad-Port BRI card.
http://www.digium.com/en/products/dig
On Tue, 2008-11-11 at 15:19 -0600, Karl Fife wrote:
> In Asterisk 1.6, there is an option to use the 'new g.711
> algorithm'.
> "Use the NEW ulaw/alaw codec's (slower, but cleaner)"
>
> By slower does this mean more 'expensive', or does it instead mean
> that there will be more algorithmic late
Mike Clark wrote:
I would like to do something like:
exten => s,1,playback(http://my.server.com/file.wav)
I tested and it does not work. It seems highly likely that someone would
already have done this one way or another. I know I could do a system
wget and then play the local file, but wante
I would like to do something like:
exten => s,1,playback(http://my.server.com/file.wav)
I tested and it does not work. It seems highly likely that someone would
already have done this one way or another. I know I could do a system
wget and then play the local file, but wanted something a bit mo
On Tue, Nov 11, 2008 at 4:19 PM, Karl Fife <[EMAIL PROTECTED]> wrote:
> In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.
> "Use the NEW ulaw/alaw codec's (slower, but cleaner)"
>
> By slower does this mean more 'expensive', or does it instead mean that
> there will be more al
In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.
"Use the NEW ulaw/alaw codec's (slower, but cleaner)"
By slower does this mean more 'expensive', or does it instead mean that there
will be more algorithmic latency? Both? Can anyone speak to the relative
increases?
With
core show channels shows all channels and the first part of the ouput
gives you the technology:
*CLI> core show channels
Channel Location State
Application(Data)
SIP/xxx (None) Up Bridged
Call(Zap/2-1)
Zap/2-1
> I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy
> is working fine but for some reason I cannot.
> The two machines have the same kernel, motherboard, the same gcc version
> and the same zaptel 1.4.8. On the second machine zaptel compiles without
> errors and ztdummy.ko is
At 11:58 11/11/2008, Dave Fullerton wrote:
>Tilghman Lesher wrote:
>> On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
>>> Jared Smith wrote:
On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
> Anyone looking for firmware should get it now before it disappears.
I remember having that issue once and I fixed it by changing a
configuration in the motherboard BIOS. It was related to the SATA hard
drive mode it was running on. The default configuration was set to
legacy mode I believe and when I changed it to enhanced mode the lost of
interrupts problem I wa
thanks a lot!
On Tue, Nov 11, 2008 at 6:06 PM, Tim Nelson <[EMAIL PROTECTED]> wrote:
> 'oh323 show channels' I would assume... I don't have a box handy with h323
> loaded to verify.
>
> Check http://astrecipes.net/index.php?n=89
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727
Tilghman Lesher wrote:
> On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
>> Jared Smith wrote:
>>> On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
Anyone looking for firmware should get it now before it disappears.
>>> It's my understanding that this isn't a fluke, but that
Hi,
I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy
is working fine but for some reason I cannot.
The two machines have the same kernel, motherboard, the same gcc version
and the same zaptel 1.4.8. On the second machine zaptel compiles without
errors and ztdummy.ko is g
On Tuesday 11 November 2008 10:50:26 Dave Fullerton wrote:
> Jared Smith wrote:
> > On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
> >> Anyone looking for firmware should get it now before it disappears.
> >
> > It's my understanding that this isn't a fluke, but that Polycom has
> > indee
On Tuesday 11 November 2008 05:17:30 Matt Riddell wrote:
> On 11/11/2008 10:48 p.m., samuel wrote:
> > So far I've updated a few machines (1.4.22) and the DNS queries are
> > reduced to a minimum, at least haven't seen DNS channel queries...
>
> Slightly more useful to me, does anyone know the patc
'oh323 show channels' I would assume... I don't have a box handy with h323
loaded to verify.
Check http://astrecipes.net/index.php?n=89
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "nik600" <[EMAIL PROTECTED]> wrote:
> And if i have an h323 configuration?
>
> Tha
Raj Jain wrote:
> On Mon, Nov 10, 2008 at 6:56 AM, Tony Mountifield
> <[EMAIL PROTECTED]> wrote:
>
>> Does anyone here know anything about GEN-GEN analogue circuits, also
>> known as Manual Ring-Down (MRD)? Apparently they are widely used in
>> Hoot'n'Holler systems for financial dealer-boards.
Well, we're running an asterisk 1.4.x system with a te220 span adapter
(T1 PRI). 83 internal SIP peers, mostly Polycom Soundpoint IP series
phones. It's a single-CPU dual-core Pentium E6420. The OS is CentOS 4.5
(x86_64). Seems to work well, though it's only busy with call switching,
voicemail,
And if i have an h323 configuration?
Thanks
On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
>
> assuming you want SIP... substitute sip for iax2 if you prefer...
>
> Tim Nelson
> Systems/Network Support
> Rockbochs I
Jared Smith wrote:
> On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
>> Anyone looking for firmware should get it now before it disappears.
>
> It's my understanding that this isn't a fluke, but that Polycom has
> indeed changed their policy and will no longer you to go through your
> res
On Tue, 2008-11-11 at 10:41 -0500, Dave Fullerton wrote:
> Anyone looking for firmware should get it now before it disappears.
It's my understanding that this isn't a fluke, but that Polycom has
indeed changed their policy and will no longer you to go through your
reseller to get the latest and gr
Dave Fullerton wrote:
> Not sure if Polycom is changing their policy or if this is an accident,
>
As far as I've seen it reported here, it's a policy change.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Li
Downloading right now, thank you very much for sharing it with us.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Convergent Technologies Core
Canall Tecnologia em Comunicações
Passo Fundo - RS - B
hi all
do you guys know why asterisk sometimes, in the cdr put the dst (the
extension) number in the src ??
I have 4 digit extensions (DID) and sometimes, the same values if found
in the src that usually have the calling user caller id.
Thanks
___
--
Hi,
You can convert your music files in 8000 hz and mono with sox command like
this
sox yourfile.wav -r 8000 -c 1 yournewfile,wav resample -ql
Uros
--
Use Free Software http://www.fsf.org/
---
Four essential software freedoms:
1) To study source code
First of all apologies for not catching up that conversation, I hadn't found
it before. Thanks for the pointer.
I wouldn't recommend using wildcards for this problem because sometimes the
DNS query contains domain names which are valid and might be used for other
actions (such as registering,mail,
Not sure if Polycom is changing their policy or if this is an accident,
but you can actually download SIP 3.1.1 right from their web site.
Anyone looking for firmware should get it now before it disappears.
SIP app and release notes can be found here:
http://www.polycom.com/usa/en/support/voice
[EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels'
assuming you want SIP... substitute sip for iax2 if you prefer...
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- "nik600" <[EMAIL PROTECTED]> wrote:
> Hi to all.
>
> Is possible with the Asterisk 1.4 cli view th
Hi to all.
Is possible with the Asterisk 1.4 cli view the current calls and their codec?
Thanks to all
--
/*/
nik600
http://www.kumbe.it
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing lis
Dear Fateme
two good refrences:
http://articles.techrepublic.com.com/2415-1035_11-94140.html
and
http://www.trixbox.org/forums/vendor-forums-certified/sangoma/solved-sangoma-101d-card-trixbox-asterisk-1-4-19-1
hope to help u
best
Pezhman
--- On Tue, 11/11/08, fateme fatah <[EMAIL PROTECTED]> w
On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote:
> Hi,
>
> I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but
> then my TE410P alarms stay RED and no zap channels can be created, even
> if they are correctly listed by "zap show channels". I tried adding
Mark Michelson wrote:
> c james wrote:
>> Mark Michelson wrote:
>>> c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include
Hello guys, I am having some problems with calls comming from the PSTN
lines, when somebody calls people can't hear me, but I can hear them, every
day I have to do a /etc/init.d/asterisk stop && /etc/init.d/dahdi restart to
have calls with sound again, wich cli dubug commands can I use to see what
is any command , shows the current rate of each channel?
--- On Mon, 11/10/08, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
From: Kristian Kielhofner <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] changing the size of voice packets
To: "Asterisk Users Mailing List - Non-Commercial Discussio
On Tue, 11 Nov 2008, Peter Evans wrote:
> On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote:
> > hii guys:
> > i get the message from the asterisk:
> >Started music on hold, class 'default', on Local/[EMAIL
> > PROTECTED],1
> > [2008-11-11 14:32:41] WARNING[1781]: fo
Matt Riddell schrieb:
> All solved by a caching dns server - achieved in debian by:
>
> apt-get install bind9
>
> followed by change nameserver to 127.0.0.1 in resolv.conf
http://lists.digium.com/pipermail/asterisk-users/2008-September/218764.html
Philipp Kempgen
--
http://www.das-aste
My suggestion is to have an agent log-off from the ACD system (or at least
pause) before attempting outbound. You can achieve that with some handy
macro that might do this transparently before the call is placed and when it
terminates.
Just my two cents,
l.
2008/11/5 Ricardo Melendez <[EMAIL P
Hi,
I'm sorry i didn't check the recipient while replying.
Sorry about the noise...
Laurent
Le 11 nov. 08 à 11:44, Laurent Caron <[EMAIL PROTECTED]> a écrit :
> Bonjour Louis-David,
>
> Asterisk envoie-t-il le signal au boitier pour le failover ?
>
> Laurent
>
>>
>>
>>
>>
>>
>>
>>
_
Hi:
I've configured an asterisk server with A102d sangoma's card and the E1 link.I
want to dial outside number using the E1 Link.How can I dial a phone number?Is
this true?
exten => 123,1,Dial(ZAP/1/phone number)
I'd appreciate any help.
___
Bonjour Louis-David,
Asterisk envoie-t-il le signal au boitier pour le failover ?
Laurent
Le 11 nov. 08 à 08:49, Louis-David Mitterrand <[EMAIL PROTECTED]
g> a écrit :
> Hi,
>
> I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22
> but
> then my TE410P alarms stay RED and
JT
Thanks for this detailed response. It's clear I have some more homework to
do before going anywhere near Mantis, but I will follow up either way.
Regards,
Richard
On Tue, Oct 28, 2008 at 9:02 PM, John Todd <[EMAIL PROTECTED]> wrote:
>
> This seems like a transcoding issue, and the RTP code m
On 11/11/2008 10:48 p.m., samuel wrote:
> So far I've updated a few machines (1.4.22) and the DNS queries are reduced
> to a minimum, at least haven't seen DNS channel queries...
Slightly more useful to me, does anyone know the patch that fixed it, so
I can update machines I don't want to upgrade
^_^
asterisk should Encoding voice in 8KHZ ,16k bits,mono8
,i have formate the .wav files.
thank you for your advice,best regard.
2008-11-11
邱磊
发件人： Peter Evans
发送时间： 2008-11-11 14:57:36
收件人： Asterisk Users Mailing List - Non-Commercial Discussion
抄送：
主题： Re: [asterisk-users] musi
thank you，Ihave get the solution。
asterisk should Encoding voice in 8KHZ ,16k bits,mono8 ^^
2008-11-11
邱磊
发件人： Lee, John (Sydney)
发送时间： 2008-11-11 14:55:00
收件人： Asterisk Users Mailing List - Non-Commercial Discussion
抄送：
主题： Re: [asterisk-users] music on hold
The reason is you
2008/11/10 Matt Riddell <[EMAIL PROTECTED]>
> On 11/11/2008 1:34 a.m., samuel wrote:
> > Folks,
> >
> > I'm tracing this error and looks like newer 1.4 (1.4.22) does not behave
> as
> > I stated (no DNS for channel domainname) and it must have been solved in
> the
> > way from my versions to the n
On Mon, Nov 10, 2008 at 07:34:34PM -0500, Jim Duda wrote:
> I've been having trouble with making outbound calls to my
> TELCO from a TDM400 card (FXS KS signalling) after upgrading
> from 1.6-beta9 to 1.6.0. The problem is completely intermittent.
>
> When it fails, I get this message:
> [Nov 9
Hi,
I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by "zap show channels". I tried adding
"dahdichanname = no" to asterisk.conf's [options] to no effect.
Going back to 1.
On Tue, Nov 11, 2008 at 05:53:39PM +1100, Lee, John (Sydney) wrote:
> The reason is your audio file is too high quality.
>
> Asterisk can only play back audio file of 4000Hz.
8000Hz (and also: mono, 16 bit sample rate).
What is the output of: file path/to/sound.wav
--
Tzafrir C
70 matches
Mail list logo
