Hi,
I need a fax-send - setup. I read the book Asterisk The Definitive
Guide chapter 19 (fax) and found 2 options listed there.
1) Using spandsp.
2) Using FFA (Digium Fax For Asterisk).
But the book nor any other article I read point out, what the
differences or drawbacks are.
Does anyone
Hi,
i am trying to install the just from git cloned app_swift version.
Compiling works fine. Install as well. But if i try to load the module
at Asterisk it fails with.
Command 'module load app_swift.so ' failed.
[Jun 20 11:29:51] WARNING[24217]: loader.c:460 load_dynamic_module:
Error
Good morning, I'm trying to distinctive ring internal/external the channel bank
FXS, after some research that has to be checked by dahdi, but I can not use
someone could tell me how should I proceed ??
Thanks
[ ]'s
Willian Castello de Alcantara
Ensite Telecom
mail: will...@ensite.com.br
Hi Jakob,
I just finished replying to your direct email (which you can disregard now as
this seems to be a different problem). I'm pretty sure I know what the issue
is, but I'll have to get back to you later this evening (my time).
- D
On Jun 20, 2012, at 4:41 AM, Jakob-Matthias Böttger
Hello,
I would do the billing on Asterisk with A2Billing, but at the configuration
of the mysql database when trying to display the table I created with this
command: mysql-u root-p mya2billing
with
mysql show tablename
since there is not even able to show tables
I then return
- And nothing else
did you put the ; at the end of the sql?
show tablename;
regards.
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Gorguez Ka
Gesendet: Mittwoch, 20. Juni 2012 14:52
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] urgent
Am 20.06.2012 14:24, schrieb Darren Sessions:
Hi Jakob,
I just finished replying to your direct email (which you can disregard now as
this seems to be a different problem). I'm pretty sure I know what the issue
is, but I'll have to get back to you later this evening (my time).
- D
On Jun
whats the output when you do this on mysql
*mysql show databases;*
If there is no database defined then you definitely need to go through the
installation steps and see if you've missed to create the A2billing
Database.
On Wed, Jun 20, 2012 at 5:59 PM, b.ti...@pinguin.ag wrote:
did you put
thank you guys, i just forgot ';' at the end of request!
thx!!!
2012/6/20 SamyGo govoi...@gmail.com
whats the output when you do this on mysql
*mysql show databases;*
If there is no database defined then you definitely need to go through the
installation steps and see if you've missed to
Thanks Lyle,
Sorry to sound so much like a newb but in asterisk I am. I was initially
trying
to do things by hand in the extensions.conf file and had no luck. I then got
from SVN checkout asterisk-gui and used it to simply try and get things
started,
and created a trunk, users, incoming
On 06/20/2012 08:44 AM, Joseph Towery wrote:
Sorry to sound so much like a newb but in asterisk I am. I was initially
trying to do things by hand in the extensions.conf file and had no luck.
I then got from SVN checkout asterisk-gui and used it to simply try and
get things started, and created
I have not use a TDM4xx card for a while, but I remember that in order
for ringing to work, you had to plug in an extra molex connector into
the card to supply power to the ringing generator portion.
If you forgot to do that...
Lyle
BTW, I know about being a noobie. I was there once myself
Kevin,
Thanks for the tip, the answer is yes, (I forgot I copy the first message in
into the body below,) but I have read a lot in the
http://cdn.oreilly.com/books/9780596510480.pdf and
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html pages. I
was
just wanting to get the
Yes, I have connected that, and the pci card has the lights on. I can now lift
the receiver on the analog phone get dial tone and dial out. Next I need to
get
the phone to ring when called. Off to do more research.
Thanks for your help.
From: Lyle Giese
On 06/20/2012 09:34 AM, Joseph Towery wrote:
Thanks for the tip, the answer is yes, (I forgot I copy the first
message in into the body below,) but I have read a lot in the
http://cdn.oreilly.com/books/9780596510480.pdf and
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html
I have two GV numbers. Both are configured to send calls to my Asterisk
1.8.13.0 box using the Google chat interface. At one time I had both
working with Asterisk. Now, for whatever reason, one of them has stopped
sending incoming calls to my asterisk box and instead just rolls to GV
voicemail.
https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE
is there some way to write userfield,accountcode to the cel?
Dne 5.6.2012 13:21, Marek Cervenka napsal(a):
hello,
is there someone who successfully get info about attended transfer
from
On Wed, Jun 20, 2012 at 11:20 AM, Chris Gentle gent...@gmail.com wrote:
I have two GV numbers. Both are configured to send calls to my Asterisk
1.8.13.0 box using the Google chat interface. At one time I had both
working with Asterisk. Now, for whatever reason, one of them has stopped
I'm with Warren... include your firewall configuration / network setup.
Andrew McRory
Sayso Communications, Inc.
2850 Industrial Plaza
Tallahassee, Florida 32301
Office) 850-224-5737
Mobile) 850-778-3206
On 6/20/2012 1:14 PM, Warren Selby wrote:
On Wed, Jun 20, 2012 at 11:20 AM, Chris Gentle
Thanks. I will go back and use that reference. I was using examples on web
pages I was trying to use and just got confused with too much information.
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Sent: Wed, June 20, 2012
On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wcse...@selbytech.comwrote:
As you said, GV and asterisk integration is unstable at best. I haven't
worked with it in a while, to be honest. But, with all that being said,
I'm not opposed to popping my GV test box back online and helping to
Hello,
1) I am wondering what is the best practice to monitor if there are or were
problems with SIP calls on my Asterisk box. E.g. how about a software that
extracts all calls from the /var/log/asterisk/full (I have permanently
enabled verbose 10 and sip debug) log and tells me on which of them
- Original Message -
Hello,
1) I am wondering what is the best practice to monitor if there are
or were problems with SIP calls on my Asterisk box. E.g. how about a
software that extracts all calls from the /var/log/asterisk/full (I
have permanently enabled verbose 10 and sip
Hello,
is there anywhere an overview of SIP error codes and under which condition
they are reported by Asterisk?
There are general definitions for SIP error codes, but they are quite
general and it's Asterisk that actually checks what's wrong and then
reports an error. Now, currently I could
- Original Message -
- Original Message -
Hello,
1) I am wondering what is the best practice to monitor if there are
or were problems with SIP calls on my Asterisk box. E.g. how about
a
software that extracts all calls from the /var/log/asterisk/full (I
have
Yeah, I noted that too, but besides that it seems like it is exactly what I
am looking for. I am especially confused that there's no hint like hey,
buy our new product, just EOL. So let's say I am looking for an
alternative to this. And unfortunately I have to add it's for private use
and I
Stefan at WPF wrote:
is there anywhere an overview of SIP error codes and under which
condition they are reported by Asterisk?
There are general definitions for SIP error codes, but they are quite
general and it's Asterisk that actually checks what's wrong and then
reports an error. Now,
Thank you Jonathan, I have read up on this, therefore 488 and the
referenced 606 error, but I have to say it wouldn't have helped me. I find
the description still very general. If one looks at the asterisk source
code, then one can clearly find the case with the crypto line and missing
RTP/SAVP.
I have a meetme running that is taking audio from a PC running asterisk
(console) as input
to my server that is then feeding it using meetme to two other asterisk
PC's going out the console.
All running 1.4.43
I have noticed that when the meetme first starts if I change the input
audio (new
On Wed, Jun 20, 2012 at 12:30 PM, Chris Gentle gent...@gmail.com wrote:
On Wed, Jun 20, 2012 at 12:14 PM, Warren Selby wcse...@selbytech.comwrote:
As you said, GV and asterisk integration is unstable at best. I haven't
worked with it in a while, to be honest. But, with all that being said,
I'm trying to set the callerid on a SIP call:
same=n,Set(CALLERID(all)=test2023214321)
same=n,Dial(SIP/home_outgoing/150)
-- Executing [202454@from-test-sip:3] Set(SIP/sip-test-0019,
CALLERID(all)=test2023214321) in new stack
-- Executing [202454@from-test-sip:4]
On Wed, Jun 20, 2012 at 3:21 PM, sean darcy seandar...@gmail.com wrote:
[home_outgoing]
type=friend
transport=tcp
secret=
fromuser=office_incoming
host=dynamic
disallow=all
allow=ulaw
It's because you're using fromuser as your username setting. This will
overwrite your CallerID
Dne 20.6.2012 18:40, Marek Cervenka napsal(a):
https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE
is there some way to write userfield,accountcode to the cel?
solved. it's set(CHANNEL(userfield)=something)
another question
i'm using
Hi,
I'm looking for someone who can help us setup Fax with T. 38 on asterisk
10.x.x - We need to be able to do FoIP (Fax over IP) as we have no pstn
lines available.
Do you know how to setup a reliable fax system, then we will pay you to
help us do this.
Regards
Thorben
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