On 11/20/06, Ralph Liebessohn <[EMAIL PROTECTED]> wrote:
On 11/20/06, Alex Robar <[EMAIL PROTECTED]> wrote:
> Hi Ralph,
>
> Have you setup your PAP2 to allow the 729 codec? I believe you actually
> have to tell it that it's allowed to use that codec before it
ere is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.
Thank you all.
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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$my_var="123"; Saydigits receives a NULL as options. And so
nothing was inserted into db.
I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed through
it directly like Joel Lansden reported on
9/14/06.
Is there another function or way to test it or I must try in anot
On 1/10/07, Lee Jenkins <[EMAIL PROTECTED]> wrote:
Ralph Liebessohn wrote:
> Hi,
>
> I'm trying to write a AGI in PHP to get the numbers dialed (with
> read()), save it into a variable to insert it into a SQL server
> database. But I cannot see results into the variabl
On 1/10/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
>From: "Ralph Liebessohn" <[EMAIL PROTECTED]>
I did a quick test and it seems that everything passed to AGI is by value,
and there is no apparent relationship between variable named used in two
different AGI commands.
How
including filename to the
script
Hope this helps
Mike
Mike,
it didn't help.
I just SOLVED the problem! You're a genius.
Now I can get information from dialplan.
Do you know why the other ways didn't work?
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
__
On 1/11/07, Ralph Liebessohn <[EMAIL PROTECTED]> wrote:
Mike,
it didn't help.
I just SOLVED the problem! You're a genius.
Now I can get information from dialplan.
Do you know why the other ways didn't work?
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
Erra
result of my_var)
All the variables here was my_var, it worked for GET VARIABLE but didn't for
SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert
my_var value into a db?
- What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for
digits.
- STDIN
GI Script Executing Application: (sayalpha) Options: (123) // Other
parameters
-- AGI Script Executing Application: (sayalpha) Options: (321)
-- AGI Script Executing Application: (sayalpha) Options: (111)
-- AGI Script Executing Application: (sayalpha) Options: (222)
AGI receives m
rsion, it will take multiple params into argv[0],
> argv[1], argv[2], etc
Eric,
I tried it on asterisk 1.2.13 and it worked with multiple params.
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Using debian you could only
# apt-get install asterisk
and it will work.
If you need I can send you a tutorial/script to install * on debian with cdr
in postgres.
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
___
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ng fax as sip client behind a PAP2.
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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: /usr/share/asterisk/sounds/ /var/lib/asterisk/sounds/
MOH: /usr/share/asterisk/mohmp3/
Logs: /var/log/asterisk/
AGIs: /var/lib/asterisk/agi-bin
Database: /var/lib/postgresql
--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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ey're the network, too.When I set everybody as pri_net this message stops.Today, I put the E1 channel to work, it was only set the channel to pri_cpe and dial !
I still without know why the previous tests didn´t work.Thanks everybody.-- Ralph Liebessohn
ICQ: 74835911Skype: liebessohn
On 7/26/06, Zenone <[EMAIL PROTECTED]> wrote:
But my question was, is it possible to free the channel if it rings toolong?MichelUsing this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls ().
-- Ra
Hi,When I am calling a queue and nobody pick the call the music on hold stop and start again.Does anybody know how to get it off and put the music on hold playing stopless until somebody pick the call? == Spawn extension (default, 12346, 1) exited non-zero on '
Local/[EMAIL PROTECTED],2' -- out
Hi guys,I am fighting to get a Wildcard TE405P working but it always start and put all channels in use. 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use)
16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use)I've tried to downgrade zaptel and asteri
On 7/24/06, Thomas Laurids Pedersen <[EMAIL PROTECTED]> wrote:
I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Lincoln Zuljewic Silva
Hello all. I have a Digium TE110P board a
On 8/11/06, Ralph Liebessohn <[EMAIL PROTECTED]> wrote:
On 7/24/06, Thomas Laurids Pedersen <[EMAIL PROTECTED]> wrote:
I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Linc
On 8/16/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume
Hi,I'm trying to start with Asterisk, but I could not put 2 softphones to talk.The asterisk server rejects the connections always when I dial.May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
192.168.0.106What is necessary to put it to work?There is no need to configure external lines.
On 5/18/06, Benchev <[EMAIL PROTECTED]> wrote:
> I'm trying to start with Asterisk, but I could not put 2 softphones to> talk. The asterisk server rejects the connections always when I dial.>> May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from
192.168.0.106>> What is necessary to put it t
On 5/18/06, Stefan Märkle <[EMAIL PROTECTED]> wrote:
Try puting apermit=0.0.0.0/0.0.0.0In the sip.conf for your two phones.BTW: your extensions.conf looks silly, you'll only be able to call test3 from test3.Busy most of the time ;-)
Stefan Märkle>Try puting a>permit=0.0.0.0/0.0.0.0>in the sip.conf
Hi,Can I, just for test, use a crossover cable linking 2 channels of my E1 card (TE406P) and dial from one channel to another?Is there any different way to do this?-- Ralph Liebessohn
ICQ: 74835911Skype: liebessohn
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On 6/20/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Tuesday 20 June 2006 11:30, Brian Swan wrote:> 3. Patience and lots of "vi zconfig.h": Try each echo canceler, with> and without the "Aggressive" option. What eventually worked for me
> was the MG2 with Aggressive cancelation.I hate to tel
On 6/26/06, Josué Conti <[EMAIL PROTECTED]> wrote:
OK Marco, irei efetuar os testes.
Se você quiser, posso lhe ajudar no forum, estou a disposição.
Assim que você criar as contas avise para podermos já ir colaborando.
Saudações
JosuéThe differences of licenses are here: https://www.nch.com.au/c
Hi guys,I need to make a configuration to test a E1 channel, so, in the same context I created two extensions:exten => 555666,1,Dial(Zap/1/5556662)exten => 5556662,1,Dial(SIP/test)
On the E1 card I linked with a cross cable the ports 1 and 2. The leds are signaling that the connection is ok.But whe
On 7/7/06, James Hawks <[EMAIL PROTECTED]> wrote:
When you dial directly you are bypassing
the zap and just dialing an internal extension. So that is probably why dialing
directly works. As far as the cross over cable between ports 1 and 2 I have
never attempted something like that
On 7/7/06, Moises Silva <[EMAIL PROTECTED]> wrote:
Oops, i missed the crossover cable part. I have used crossover cable,so it should work, but the DNID must be complete. Wich signaling areyou using?RegardsHi Moises,I'm signalling=pri_net.
I got this error too:app_dial.c: Unable to create channel o
On 9/20/06, C F <[EMAIL PROTECTED]> wrote:
Erik is this for a Mediatrix 1204? If so where did you get thesesettings? In SNMP? or HTTP?>From the Mediatrix documentation:Page 59 (87) These are footnotes to whereever the words registerserver are mentioned in the Manual:
1. The Mediatrix 1204 does not
On 10/10/06, George Masgras <[EMAIL PROTECTED]> wrote:
Hello all! I'm currently using Asterisk in conjunction with a2billing andeverything seems to be working great so far. Now, all I'm missing issome sort of a GUI to monitor all calls going out through my trunks. I
can always do 'sip show channel
On 10/17/06, Mike Clark <[EMAIL PROTECTED]> wrote:
We have several sites in this configuration with no nightly reboots. Allsites except one are problem free. One site still has dropped calls.None of the sites crashes and some of them have been up for a few weeks.
Tom Vile wrote:> fine for me here s
On 10/17/06, Thirumal Saminathan <[EMAIL PROTECTED]> wrote:
hi all,
please any one help me ,how to configure chanspy application .
and also send me if u have any sample configure file.
-thiruHi,It could be very simple, like:exten => 123,1,ChanSpy(); Spy all channelsor more accuracy:exten =>124,
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