CF,
Adding www after Dial doesnt solve the
trouble.
I think we are talking the same but I dont
express correctly.
Did you saw my dialplan? I dont 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 doesnt give
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 =
I think still didnt 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 doesnt
Ok Ok, the figure doesnt help.Here we go again - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | | Ext1 Ext2Here is my dialplan[incoming]exten = s,1,Answerexten = s,2,Background(prueba-pbx)exten =
Ok,
Im 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
Hello,
Ive got asterisk running and almost working
with Panasonic KX-TD1232
I said almost, because theres a strange behaviour
when I make calls.
---
-
-
---
| SIP | -- | ASTERISK | -- | PANASONIC
| | PSTN
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 doesnt
Really dont.
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/1002|20
I really dont understand what you say.
Ive been searching in my SIP device (Innomedia
3308), and there isnt any option to disable 3-way calling. Do you refer
to sip.conf???
Pablo
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Steven,
Ive been searching that you say, but certainly
I dont know where to search or those lines isnt there.
I found these:
Configuring VoIP DigitMap
dialing pattern
- empty -
Configure FXS Setting Parameters
Ringing Timeout = 180 second
Ringing Cadence = 0
Ringing
Not sure what you want, but I have asterisk running
on Centos 4.3 and theres no problems.
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Hello,
Ive 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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