On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones allan...@gmail.com wrote:
But jailbreaking increases the freedom to develop a application and
Oh, I agree with you, but it's probably even better to make a decision
to either buy into the constraints of Apple or find a better, free-er
phone, which is
Hi,
But the caller ID function is still not working my system.
Please Help.
Thanks,
Arun S
On Wed, Jan 6, 2010 at 11:13 AM, Kyle Kienapfel doctor.w...@gmail.comwrote:
On Tue, Jan 5, 2010 at 5:24 AM, Arun Sasidhar
arun.sasid...@cabotsolutions.com wrote:
Hi,
I am using asterisknow
Hello
I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started the
first example Hello AGI in this tutorial
http://asterisk-java.org/development/tutorial.html I put the jar file and
the proparty file in folder
i wrote in extensions.conf this exten = 1300,1,AGI(agi://
Hi,
You can directly call that class like AGI(com.abc.cde.Hello) . Hello is
class name.
Hope this helps
On Wed, Jan 6, 2010 at 2:16 PM, ahmed magdy amagdy.ibra...@gmail.comwrote:
Hello
I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started
the first example Hello AGI in
Hello,
can any of the Asterisk API be used to dial a number on MITEL telephones?
Many thanks.
phiroc
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Dear All
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
I have defined the following profile for my sip phone :
Under sip.conf :
-
[osaka]
type=friend
context=sip-outgoing
host=192.168.0.139
disallow=all
On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing Unknown when there is an incoming call. I think the
You can try this
[agi_test]
exten = 123,1,Answer();
exten = 123,n,noop(${CALLERID(num)})
exten = 123,n,set(IP_FOR_AGI=192.168.127.58)
exten =123,n,Agi(agi://${IP_FOR_AGI}/com.package.ClassName)
On Wed, Jan 6, 2010 at 2:28 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hi,
You can
Hi all,
I need Help. I want to compile zaptel in data mode but i got this errors:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618: error: implicit
declaration of function âhdlc_statsâ
Hi,
I noticed you always prefix 'Inquiry:' to your questions on the list.
This is implied from the subject line itself, and wastes some space in
the subject line, so I guess it is kind of pointless.
Now to the question itself,
On Wed, Jan 06, 2010 at 10:44:31AM +, hadi motamedi wrote:
Can
On Wed, Jan 6, 2010 at 11:55 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Hi,
I noticed you always prefix 'Inquiry:' to your questions on the list.
This is implied from the subject line itself, and wastes some space in
the subject line, so I guess it is kind of pointless.
Now to the
Hi,
I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*
On Wed, Jan 6, 2010 at 4:40 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jan 05, 2010 at 06:54:18PM +0530, Arun Sasidhar wrote:
Hi,
I am
On Wed, 6 Jan 2010, Arun Sasidhar wrote:
Hi,
I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*
Hi Arun,
Just for fun I read over the bug id you quoted below, and it seems there
are a number of settings you may
Nicholas,
Sorry I don't know, but are your calls working okay ?
Depending on the verbosity level being set, I see warning
msgs all the time, that I ignore.
Frequently, an upgrade to the next release of the same
major version also eliminates the warning msgs.
If you are really concerned,
Another thing to consider (hopefully not covered in one of the 100 previous
replies) - Callerid is a hit and miss proposition; to get ABC Widgets
205-555-1212, The Telco has to have ABC Widgets in their database and
publish it. If not, you get Unknown 205-555-1212. Since a good chunk
of
Are you even paying for the service?
Here in the US, on PSTN lines from the ILEC's, CallerID is a pay
service, with 2 tiers. Number only, and number with name.
Some CLEC's include this without extra charge, as do most/all VOIP
providers.
Do you have a box or phone, independent of the Asterisk
Please can someone help me with my queue-problems ?
When there is a member available for answering calls that entered the
queue, the caller does not hear the musiconhold as defined
musicclass=default.
What happens is that when a second caller enters the queue, the caller
hears the ringtone for
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
busydetect=yes
busycount=6
callprogress=yes
Dear,
there is a problem in codec translation..so change the ulaw codec to
g729. .if problem persist then u must have same codex on asterisk server and
clients (skype)...
On Mon, Jan 4, 2010 at 11:24 AM, Tim Panton t...@westhawk.co.uk wrote:
On 30 Dec 2009, at 19:43,
Allann-
On Wed, Jan 6, 2010 at 8:50 AM, Allann Jones allan...@gmail.com wrote:
But jailbreaking increases the freedom to develop a application and
Oh, I agree with you, but it's probably even better to make a decision
to either buy into the constraints of Apple or find a better, free-er
At 05:31 PM 1/5/2010, you wrote:
Ah, good idea. :-) Are you saying that if I got a number that was
in my parents area code then they could be making a local call to my
Asterisk, which is physically a 1000+ miles from them? Now that is
cool.
Once you have Asterisk set up you can essentially
At 07:00 AM 1/6/2010, you wrote:
Its a free service here and My ordinary phone displaying the
Caller ID without any problem.
I have done some modifications in zapata.conf
Now it looks like this
Make sure that there is between 1 and 5 seconds after the first ring
before you answer the call.
On Wed, Jan 06, 2010 at 08:30:48PM +0530, Arun Sasidhar wrote:
Hi,
Its a free service here and My ordinary phone displaying the Caller ID
without any problem.
I have done some modifications in zapata.conf
Now it looks like this
*[channels]
language=en
hanguponpolarityswitch=yes
We have recently pulled an ancient Fujitsu-branded Centigram voicemail
system out of production use and replaced it with an Asterisk box, which
is now serving as our enterprise voicemail system and automated attendant.
The Asterisk system is connected to a Fujitsu F9600 PBX and uses the
1.6.1.x
Cool - thanks!
On Wed, Jan 6, 2010 at 12:14 PM, Ira i...@extrasensory.com wrote:
At 05:31 PM 1/5/2010, you wrote:
Ah, good idea. :-) Are you saying that if I got a number that was
in my parents area code then they could be making a local call to my
Asterisk, which is physically a 1000+ miles
I fixed it, the problem was that the 2nd T1 didn't have a Switch Identifyer
set.
I set the Switch Identifyer and now I can route calls to the PSTN. Merlin
has a default that a trunk with a Null Switch Identifyer is considered a CO
trunk. So the Merlin was getting confused, and routing it to the
There is a line like in codes/Makefile
$(if $(filter
codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
What is filter? Where is filter?
whereis filter doesnt return anything
find . | grep filter in asterisk root directory returns nothing.
Thanks,
Jerry
On Wednesday 06 January 2010 13:45:55 Jerry Geis wrote:
There is a line like in codes/Makefile
$(if $(filter
codec_lpc10,$(EMBEDDED_MODS)),modules.link,codec_lpc10.so): $(LIBLPC10)
What is filter? Where is filter?
whereis filter doesnt return anything
find . | grep filter in asterisk root
It's a Makefile command. See:
http://www.gnu.org/software/automake/manual/make/Text-Functions.html#index-filter-554
great - thanks
is there no method by the configure command to --disable-FEATURE???
the help says its there but doesnt seem to do anything for me.
example: ./configure
Jerry Geis wrote:
is there no method by the configure command to --disable-FEATURE???
There is not. The Asterisk configure script is used for platform
specific settings, locating libraries and header files and the like. It
is not used (directly) for controlling which portions of Asterisk are
Hi
We have an operator that his device state on all queues is In use where it
should be Not in use.
how can we manually change the state of a device?
I looked into the devstate function and tryed the following:
perfpbxr*CLI devstate list
perfpbxr*CLI
Hi,
I need to install (within the next couple of hours) a 1.6.1.11 server with a
Digium B410P board.
One of this system's DID is dedicated to Fax reception (I don't need to send
faxes).
As I'm a bit familiar with it, I would like to use spandsp and ReceiveFAX
application to enable this feature.
Hi,
I'm sending twice the same file using Hylafax's sendfax app.
The first time I'm dialing a DID attached to the ReceiveFAX application.
The second time I'm dialing an internal extension attached to the same
ReceiveFAX application :
1. sendfax/hylafax/iaxmodem asterisk dadhi
I download the x-lite software for windows.
Put it on two laptops.
Using asterisk 1.4.28
set videosupport=yes in sip.conf general.
disallow=all
allow=h264
allow=ulaw
allow=alaw
videosupport=yes
have the above for both softphone defintions. I heard audio but no video
when I call.
I click the
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the same
ReceiveFAX application :
2. sendfax/hylafax/iaxmodem asterisk spandsp
In the 2nd case, I've got 3 craches out of 3 attempts (with a rough
hi,
i use agi to send message back to Asterisk by STDERR, but why i
could't see the message in asterisk CLI?
i start asterisk use asterisk -vc in order to see all message.
Thanks
--
Best regards,
Sucan
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On Thu, 7 Jan 2010, Zhang Shukun wrote:
i use agi to send message back to Asterisk by STDERR, but why i could't
see the message in asterisk CLI?
Output to STDERR does nothing for me either.
I prefer to use syslog() to log the messages via syslogd.
--
Thanks in advance,
I think that's very wise advice. To offer a commercial perspective, our
customers willing to pay for sophisticated
smart phone apps (currently gov/mil agencies and some mid-size telecoms)
have very specific needs and care about reliable operation, development
duration, long-term support --
Hi,I want to know how to do to work PLC of Asterisk.
Anyone plz help me.
PLC (Packet Loss Concealment) is included in Asterisk,I read at voip-info.org
or release note.
And I see in codecs.conf, genelicplc setting.
So I put codecs.conf in '/etc/asterisk' ,and wrote genericplc = true.
And I
hi,
I made changes in zapata.conf but no result.
I tried different settings. I am getting differnt logs But no result
when i use cidstart=ring
I am getting this in my asterisk log
[Jan 7 09:31:13] VERBOSE[7129] logger.c: -- Starting simple switch on
'DAHDI/1-1'
[Jan 7 09:31:14] ERROR[7129]
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
Hi,
Having problems with getting either RxFax or FaxReceive
to compile. Running Asterisk 1.4 on CentOS 5.
What version of SpanDSP do you use?
spandsp-0.0.6pre12.tgz
and:
Hello,
Maybe there is the easiest way to compile additional my module without
recompiling all asterisk?
Thanks
--
Pagarbiai / Best Regards,
Giedrius
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asterisk-users mailing
I have it running on * 1.4.19. You can get it on the internet, .so and an
intaller that checks for dependencies.
--Mensaje original--
De: Doug
Remitente: asterisk-users-boun...@lists.digium.com
Para: asterisk-users@lists.digium.com
Responder a: Asterisk Users Mailing List -
On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
hi,
I made changes in zapata.conf but no result.
You use zapata.conf . I suppose you use asterisk 1.4 . Give asterisk
1.6.0 or newer a shot.
--
Tzafrir Cohen
icq#16849755
On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
Hi,
Having problems with getting either RxFax or FaxReceive
to compile. Running Asterisk 1.4 on CentOS 5.
What version of SpanDSP
On Thu, Jan 07, 2010 at 08:08:07AM +0200, Giedrius Augys wrote:
Hello,
Maybe there is the easiest way to compile additional my module without
recompiling all asterisk?
'make' if you already have a fully-built source tree.
Optionally manual copying instead of 'make install'
If you want to
hi arun can you paste a dialplan here
and version of asterisk
regards
dhaval
On Thu, Jan 7, 2010 at 11:51 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jan 07, 2010 at 09:54:09AM +0530, Arun Sasidhar wrote:
hi,
I made changes in zapata.conf but no result.
You use
At 00:22 1/7/2010, Tzafrir Cohen wrote:
On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
Hi,
Having problems with getting either RxFax or FaxReceive
to compile. Running
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