Hello,
I own one of these netbooks Asus Eee PC 900, mine is running FreeBSD 7.0, and a
Linux based cellphone, the OpenMoko Freerunner.
Since some time I'm thinking in a replacement of my 'normal' BRI phone
at home and the two items mentioned above let me think that the
replacement should be
You could simply have it Dial() to wherever it needs to go at the end of
the script.
2009/1/6 Rajkumar S rajkum...@gmail.com
Hi,
I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does
core show function SIPPEER
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
since 1.4 you can also use
setvar=foo=bar
in sip.conf when configuring the peer. Then the channel variable foo is
automatically set to bar for calls initiated by this peer.
regards
klaus
Philipp Kempgen
Xorcom had something, usb bri, but it is pricey. If you dont need to change
provider and planning to stay with bri, why dont you buy another bri phone ?
2009/1/7 Matthias Apitz g...@unixarea.de
Hello,
I own one of these netbooks Asus Eee PC 900, mine is running FreeBSD 7.0,
and a
Linux
El día Wednesday, January 07, 2009 a las 09:50:06AM +0100, Grygoriy
Dobrovolskyy escribió:
Xorcom had something, usb bri, but it is pricey. If you dont need to change
provider and planning to stay with bri, why dont you buy another bri phone ?
Because since I own the Moko I know *what* it
On 7 Jan 2009, at 09:07, Matthias Apitz wrote:
You can SSH to it and change/install/write-by-your-own whatever you
want or what you feel missing. Now I want to have a UNIX
based bri phone. That's it, not more, but not less :-)
Perhaps you could write the driver for one? ;)
Hi, all. I just tried to fire up app_txfax and app_rxfax, only to find
that I can't seem to compile them. The problem appears to be that my
libtiff library is wrong. Only problem is that, according to the README,
I need libtiff =3.8 and 4.0, which is all well and good... except that
there is no
On Wed, Jan 07, 2009 at 05:47:18AM -0500, Ken D'Ambrosio wrote:
Hi, all. I just tried to fire up app_txfax and app_rxfax, only to find
that I can't seem to compile them. The problem appears to be that my
libtiff library is wrong. Only problem is that, according to the README,
I need libtiff
Hi,
Thanks for your reply
Can you suggest me how can we avoid it by doing any configuration changes in
asterisk.
So the freeze issue may not be occurred again!
Please provide me some help!!!
Thanks in advance!
Thanks,
Max Alex
Voip Developer
On Wed, Jan 7, 2009 at 12:58 PM, Grey Man
Hi Friends,
Currently i am using the asterisk 1.4.x version. In that i want to enable
to silence suppression in the SIP calls. Please tell me the configuration
changes to be done.
Thanks in advance,
balasam.
___
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El día Wednesday, January 07, 2009 a las 09:17:43AM +, Steve Howes escribió:
On 7 Jan 2009, at 09:07, Matthias Apitz wrote:
You can SSH to it and change/install/write-by-your-own whatever you
want or what you feel missing. Now I want to have a UNIX
based bri phone. That's it, not
There is also type=[user|peer|friend] in chan_iax and chan_h323
there is also type=h323|alias in chan_h323
maybe it is better to use in users.conf another variable, e.g.
siptype=
or
h323type=
regards
klaus
Tzafrir Cohen schrieb:
On Tue, Dec 23, 2008 at 10:35:19AM +0100, Klaus
2009/1/7 Max Alex max.aster...@gmail.com
Hi,
Thanks for your reply
Can you suggest me how can we avoid it by doing any configuration changes
in asterisk.
So the freeze issue may not be occurred again!
Please provide me some help!!!
Thanks in advance!
Thanks,
Max Alex
Voip Developer
I have a smartnet contract for this phone, and have searched high and
low for this file on the Cisco website.
I need:
United_States/7960-tones.xml
English_United_States/7960-font.xml
Every road seems to lead to the Call manager express downloads... I
don't have a CME, so that's basically
Is it ready for prime time? I am about to install a new server that will
be processing about 3M minutes per month and running a custom AGI program
for prepaid calling cards. Need to choose between 1.4x and 1.6...
Cheers,
j
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Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow my dialplan:
exten =
Jeff LaCoursiere schrieb:
Is it ready for prime time?
He Jeff,
at least version 1.6.0-beta9 was not yet very stable.
We are also used to handle serveral Mmin/month with
asterisk 1.4, but in our test environment, our asterisk
1.6.0-beta9 consumed file handles without releasing,
and even a
http://bugs.digium.com/view.php?id=14188
regards
klaus
Klaus Darilion schrieb:
There is also type=[user|peer|friend] in chan_iax and chan_h323
there is also type=h323|alias in chan_h323
maybe it is better to use in users.conf another variable, e.g.
siptype=
or
h323type=
Daniel Varella wrote:
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow
Jeff LaCoursiere j...@jeff.net writes:
Is it ready for prime time? I am about to install a new server that will
be processing about 3M minutes per month and running a custom AGI program
for prepaid calling cards. Need to choose between 1.4x and 1.6...
1.6.0.x is reasonably ok. We have it
As much as I'm an open source guy, but the OpenMoko phones are worthless
IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it
too much to ask for at least EDGE if not 3G?
Matthias Apitz wrote:
El día Wednesday, January 07, 2009 a las 09:50:06AM +0100, Grygoriy
Daniel Varella a écrit :
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow
What version of spandsp do you use?
Based on the fact that you asked that question, I suddenly got suspicious
that, despite his warnings, it might have worked for you with libtiff-4.
So I went and re-tried (using spandsp 0.0.4-pre16), and it failed
*differently*. So then I got suspicious that
El día Wednesday, January 07, 2009 a las 10:55:55AM -0500, Singer XJ Wang
escribió:
As much as I'm an open source guy, but the OpenMoko phones are worthless
IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it
too much to ask for at least EDGE if not 3G?
We go offtopic of
On Wed, 7 Jan 2009, Matthias Apitz wrote:
El d?a Wednesday, January 07, 2009 a las 10:55:55AM -0500, Singer XJ Wang
escribi?:
As much as I'm an open source guy, but the OpenMoko phones are worthless
IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it
too much to ask
Try
module reload pbx_lua.so
Dominique Dartois wrote:
Hello all.
I'm playing with LUA and I can't see a way to reload 'extensions.lua' after
a change, except by restarting Asterisk.
Any clue?
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Mikel,
On Thu, Jan 08, 2009 at 12:52:02AM +1100, Mikel Lindsaar wrote:
I have a smartnet contract for this phone, and have searched high and
low for this file on the Cisco website.
I need:
United_States/7960-tones.xml
English_United_States/7960-font.xml
Every road seems to lead to the
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
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As much as I'm an open source guy, I'm also big on making my life
easier. I use to bottom post till I got a BlackBerry. With BB (and also
Treo, Windows Mobile, iPhones) its just easier to read with top posting.
The phones generally download first X (I belive X = 4 for the
BlackBerry) KB of
I recently did an upgrade from 1.4.18 to 1.4.22 and now I am having a
problem with chan_alsa. It seems to work for a while but after a few pages I
start getting this error:
[Jan 7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable
This
Grygoriy Dobrovolskyy schrieb:
core show function SIPPEER
Does not work. Using the SIPPEER function you have to know the name of
the peer already.
regards
klaus
2009/1/6 Klaus Darilion klaus.mailingli...@pernau.at
mailto:klaus.mailingli...@pernau.at
since 1.4 you can also use
Ken D'Ambrosio wrote:
Based on the fact that you asked that question, I suddenly got suspicious
that, despite his warnings, it might have worked for you with libtiff-4.
So I went and re-tried (using spandsp 0.0.4-pre16), and it failed
*differently*. So then I got suspicious that my previous
I can do a great Colonel Klink and pretty good Shulz imitation...in case you
want me to record some prompts.
:)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: January 7, 2009 11:31 AM
On Wed, Jan 07, 2009 at 10:55:55AM -0500, Singer XJ Wang wrote:
As much as I'm an open source guy, but the OpenMoko phones are worthless
IMHO. Its suppose to be a modern phone, but seriously GRPS only? Is it
too much to ask for at least EDGE if not 3G?
Irrelevant for this use case: for your
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
/usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9,
from
Hi,
URL http://lists.beronet.com/mailman/listinfo/misdn-asterisk in
http://www.misdn.org/index.php/Support page returns Not Found.
Is this list still active ?
Regards
___
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2009/1/7 Olivier oza-4...@myamail.com
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
/usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9,
from
On Wed, Jan 07, 2009 at 06:19:42PM +0100, Olivier wrote:
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
/usr/src/mISDN-1_1_8/drivers/isdn/hardware/mISDN/core.h:9,
You know NOTHING!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of OCG Technical Support
Sent: Wednesday, January 07, 2009 11:54 AM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] recommendation
The continuing discussion on a future Simple CDR mode of generation...
from which I will extract the info and add a section to the
overall CDR spec I'm developing...
For newcomers, Simple CDR mode would not break the conversations
into legs at all. Each CDR would simply record the total time
Great!!
Thanks a lot.
Try
module reload pbx_lua.so
Dominique Dartois wrote:
Hello all.
I'm playing with LUA and I can't see a way to reload 'extensions.lua'
after a change, except by restarting Asterisk.
Any clue?
---
Dominique Dartois
___
TianLun,
I should have know you would have wanted a Blackberry SIP client to connect
to an Asterisk box. Sorry my bad!
I knew there was a reason why I didn't choose Truphone as my SIP client.
I have an iPhone and I am currently using Fring which is local client that
connects to my Asterisk box
Hi list,
I see their is ExtenSpy(), I want to monitor calls (in and out I hope) from
another phone, all the channels are SIP. ChanSpy() looks like you cannot
give it a context and I want to be able to only monitor certain calls. Any
Ideals on how to do this?
Thanks!
The Blackberry community has been begging for a SIP client for awhile.
Apparently there are some restrictions within the Blackberry OS. But
with the newer Blackberry models including wifi abilities, we should be
seeing something released soon... I hope! **Fingers Crossed**
Eric Moniz wrote:
Does your fring work over the 3G network also or just the wifi?
Cheers,
j
On Wed, 7 Jan 2009, Eric Moniz wrote:
TianLun,
I should have know you would have wanted a Blackberry SIP client to connect
to an Asterisk box. Sorry my bad!
I knew there was a reason why I didn't choose Truphone as
Which release of * are you trying to connect to? 1.6 has Cell capability
and the skinny option is available on 1.4 and 1.6. Just a thought.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Broyles
Sent:
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
Mark Willis
___
-- Bandwidth
Actually, I see CounterPath (X Lite) has another SIP Client for BlackBerry.
However, they told me they sell that product with their wireless solution
only, not separately
On Wed, Jan 7, 2009 at 4:16 PM, Jeff LaCoursiere j...@jeff.net wrote:
Does your fring work over the 3G network also or just
Hi!
I'm part of two mailinglists. I haven't heard much from one, but in the
other there are some mails now and then. isdn4linux and then there's a
specific misdn mailinglist (I think in connection with asterisk). I'll check
my address book if you're interested in more info.
Kindest
Hey Jeff,
WiFI only for the voice app. The IM works over wifi, edge and 3G.
ATT want their $$$.
E.
On Wed, Jan 7, 2009 at 4:16 PM, Jeff LaCoursiere j...@jeff.net wrote:
Does your fring work over the 3G network also or just the wifi?
Cheers,
j
On Wed, 7 Jan 2009, Eric Moniz wrote:
Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients.
Could be what you want.
Greets!
Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz:
TianLun,
I should have know you would have wanted a Blackberry SIP client to
connect to an Asterisk box. Sorry my bad!
I knew
Hi Mark -
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
You really want to do SLA with all 23 lines of the PRI? That's a
I went from asterisk 1.4.22 (which was working) to SVN
and I am getting the message rejected because extension not found...
How can I modify the print statement in chan_sip.c line 18388 to include
not
just the extension but the context its trying to find my extension in???
From the product description, i dont think Gizmo5 allows me to register the
client with my asterisk. If i am wrong, please let me know
On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo
rodolfo.alca...@padep.org.bo wrote:
Missed the thread, sorry. Gizmo5.com has some blackberry SIP
2009/1/7 TianLun Song stl...@gmail.com
From the product description, i dont think Gizmo5 allows me to register the
client with my asterisk. If i am wrong, please let me know
On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo
rodolfo.alca...@padep.org.bo wrote:
Missed the thread,
On Wednesday 07 January 2009 15:53:46 Jerry Geis wrote:
I went from asterisk 1.4.22 (which was working) to SVN
and I am getting the message rejected because extension not found...
How can I modify the print statement in chan_sip.c line 18388 to include
not
just the extension but the context
2009/1/7 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Jan 07, 2009 at 06:19:42PM +0100, Olivier wrote:
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
You really want to do SLA with all 23 lines of the PRI? That's a
Ron McCarthy wrote:
Hi list,
I see their is ExtenSpy(), I want to monitor calls (in and out I hope)
from another phone, all the channels are SIP. ChanSpy() looks like you
cannot give it a context and I want to be able to only monitor certain
calls. Any Ideals on how to do this?
Hi
Looks like it was it. Now it works OK. Thanks for help
Cheers
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
HI... I found that there is a cmd called, bridge in 1.6. Does it work
what I expect? How can I use it if it works as I can't find any
information about it in web?
On Tue, Jan 6, 2009 at 6:18 PM, Nick Wolf new...@gmail.com wrote:
I am also interested in establishing a three way conversation
Thanks Mark,
The phone starts, I can get to settings and such... just it keeps
looking for this file.
I just opened a TAC ticket and getting it handled.
Mikel
On Thu, Jan 8, 2009 at 3:26 AM, Mark G. Thomas m...@misty.com wrote:
Mikel,
On Thu, Jan 08, 2009 at 12:52:02AM +1100, Mikel
Is there a method or can there be a method added
to dahdi to only compile for dadhi_dummy?
No other modules are needed sometimes.
no firmware downloads are needed.
Thanks,
Jerry
___
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Hi Mark -
You really want to do SLA with all 23 lines of the PRI? That's a
lotta lines to be shared. You'd need two sidecars for each phone
(Cisco or Polycom).
Actually there will be multiple PRI's :)
This customer is a multi-tenant situation so each tenant will have a few
trunk SLA's
Works like a champ. I have to use the b option as well otherwise it just
goes into a endless beep, sounds good though!
Thanks for the help!
On Wed, Jan 7, 2009 at 4:02 PM, Mark Michelson mmichel...@digium.comwrote:
Ron McCarthy wrote:
Hi list,
I see their is ExtenSpy(), I want to monitor
On Wed, Jan 7, 2009 at 10:49 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
Tzafrir Cohen wrote:
On Tue, Jan 06, 2009 at 09:39:36PM -0600, Alejandro Kauffmann wrote:
Tzafrir Cohen wrote:
On Tue, Jan 06, 2009 at 02:28:53AM -0600, Alejandro Kauffmann wrote:
I've built SVN-trunk-r167180 and try to start it with:
asterisk -f -C /etc/asterisk/asterisk.conf
which
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