CF,
Adding www after Dial doesn’t solve the
trouble.
I think we are talking the same but I don’t
express correctly.
Did you saw my dialplan? I don’t think I would
have to add r.
Yes, I have installed a 4 FXO Card, with fxsks
signalling. What I mean is I understand FXO doesn’
Ok
Here goes dialplan
[general]
static=yes
writeprotect=yes
[incoming]
exten => s,1,Answer
exten => s,2,Background(pbx)
exten => s,3,Set(TIMEOUT(response)=5)
exten => 1001,1,Dial,SIP/1001|20
exten => 1001,2,Hangup
exten => 1001,102,Congestion,3
exten => 1002,1,Dial,
Hello,
I’ve asterisk installed, but It has a
particularity.
When I dial an extension in context internal (exten
=> 1001,1,Dial,SIP/1001), the call finishes successfully. But, when I dial
to get a trunk line (trunk like an analog line from PSTN) it takes above 15 sec
to give me tone
Pablo Mora, Ing.
GERENTE DE OPERACIONES
ESPOLTEL S.A.
Malecón 100 y Loja
Telf.:2514477
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Not sure what you want, but I have asterisk running
on Centos 4.3 and there’s no problems.
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Hi all,
Iv’ got a problem taking lines to call from SIP
to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to
wait above 15 seconds.
[out]
exten => 9,1,Dial,Zap/g1/9
exten => 9,2,Hangup
exten => 9,102,Congestion
The problem occurs when the user does
Really don’t.
Dialplan is very simple, please take a look
[incoming]
exten => s,1,Answer
exten => s,2,Background(prueba-pbx)
exten => s,3,Set(TIMEOUT(response)=5)
exten => 1001,1,Dial,SIP/1001|20
exten => 1001,2,Hangup
exten => 1001,102,Congestion,3
exten => 1002,1,Dial,SIP/1
I really don’t understand what you say.
I’ve been searching in my SIP device (Innomedia
3308), and there isn’t any option to disable 3-way calling. Do you refer
to sip.conf???
Pablo
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Steven,
I’ve been searching that you say, but certainly
I don’t know where to search or those lines isn’t there.
I found these:
Configuring VoIP DigitMap
dialing pattern
- empty -
Configure FXS Setting Parameters
Ringing Timeout = 180 seco
Hello,
I’ve got asterisk running and almost working
with Panasonic KX-TD1232
I said almost, because there’s a strange behaviour
when I make calls.
---
-
-
---
| SIP | -- | ASTERISK | -
Ok Ok, the figure doesn’t help. Here we go again… - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | |
Ok,
I’m going to stop pictures
I have a Digium 4 FXO Card in my asterisk, and
connect to Panasonic through 2 extensions (configured in a pool)
This means when you dial 200 (example) in Panasonic,
the call goes to asterisk and it answers.
In this sense, the answer is yes… rep
I think still didn’t explain me clearly…
The problem is when I dial 0, in this case the asterisk
take Zap (connected directly to ext 200 from Panasonic), Panasonic gives tone,
dial another extension (ie 100), the extension rings but when answer the phone asterisk
keeps ringing… it doesn
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