/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
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Ralph Liebessohn
ICQ: 74835911
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as sip client behind a PAP2.
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Ralph Liebessohn
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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 shoudn't get the result of READ or GET VARIABLE? Where these values
go?
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Ralph Liebessohn
ICQ: 74835911
Skype
: (111)
-- AGI Script Executing Application: (sayalpha) Options: (222)
AGI receives more than 1 parameter.
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, 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.
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asterisk
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.
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Ralph Liebessohn
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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.
However, a small adaption of dial
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?
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ICQ: 74835911
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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
Errata.
IT just SOLVED..
Not I
1.2.13.
Thank you all.
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through
it directly like Joel Lansden Joel AT digitalparadise DOT net reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?
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Ralph Liebessohn
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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 variable, it always return
NULL.
Here
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 will work.
Cheers,
Alex
On 11/20/06
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
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
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 channels'
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 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
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 Lincoln Zuljewic
Silva
Hello all
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 and when I do a 'dial
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 asterisk but it didn't solve the
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' --
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 ().
--
, 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
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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 when
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, butthe 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 of
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:
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 tell you
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 MärkleTry puting apermit=0.0.0.0/0.0.0.0in the sip.conf for
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
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 to work?
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