Il Neofita wrote:
> Hi,
> I update from asterisk 1.2 to 1.4 and I have some problems.
> In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a
> call from an external providers
> now in 1.4 I recieve only one ring
> What can I do to solve this problem?
You can start by removing the
Il Neofita wrote:
> In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a
> call from an external providers
Remove the r
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
__
Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a call
from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
___
--Bandwidth an
Thank you I will try tonight
On 9/10/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
>
> Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
> > On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]>
> > wrote:
> > Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofit
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
> On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]>
> wrote:
> Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
>
> Well, it seems there are differences between those accounts
> then.
>
On 9/9/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
>
> Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
>
> Well, it seems there are differences between those accounts then.
>
> You might want to post your sip.conf, and -if that is possible- the ATA
> conf file; or at least
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
> Hi,
> my ATA has two phones attached and the possibility to set different
> accounts.
> I put two account of my asterisk server, however, it is able to call
> only with the second one in order to the sip.conf and the first it
> gives me
Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call only
with the second one in order to the sip.conf and the first it gives me 403.
And idea how to solve it?
_
On Fri, 2007-08-31 at 11:38 -0500, Carlos Chavez wrote:
> I am having a strange problem with an Asterisk server that has a small
> 5 seat call center. While everything seems to be working properly I if
> do a "core show channels" the server goes into a loop:
I'm not sure what might cause th
I am having a strange problem with an Asterisk server that has a small
5 seat call center. While everything seems to be working properly I if
do a "core show channels" the server goes into a loop:
pbxinsol*CLI> core show channels
Channel Location State
Applicatio
Check the value of DIALSTATUS then decide of you want to dial the 2nd
number. See [macro-std-exten] in extensions.conf for an example of
checking the value of DIALSTATUS.
The only time you might want two Dial lines in a row is if you always,
not matter what, want to dial the 2nd number.
Il
Yes, but I would like to try a number and after to try a second one.
Any Idea how to avoid this.
On 2/18/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
C F wrote:
> Asterisk supports this directly by issuing the hangup command before
> the answer command. However, when using an analog i
C F wrote:
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog interface like FXO
the line has no way of knowing you just hung up and will continue to
ring, which asterisk will see as a new call. in my experience even
when using
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog interface like FXO
the line has no way of knowing you just hung up and will continue to
ring, which asterisk will see as a new call. in my experience even
when using a PRI if i d
On 15 Feb 2007, at 09:55, Yuan LIU wrote:
From: "Il Neofita" <[EMAIL PROTECTED]>
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however,
is not the case and a really did not catch why.
Now I see where the confusion comes from. Asterisk doesn'
Ok thank you a lot!!!
On 2/15/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
>From: "Il Neofita" <[EMAIL PROTECTED]>
>Date: Thu, 15 Feb 2007 03:37:14 -0500
>
>But I tought that hangup was suppose to close the call, however, is not
the
>case and a really did not catch why.
Now I see where the confusio
From: "Il Neofita" <[EMAIL PROTECTED]>
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however, is not the
case and a really did not catch why.
Now I see where the confusion comes from. Asterisk doesn't really speak
English - or Chinese for that
On 2/14/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
Well, you'll have to decide how you want to "hang up" the caller: Do you
want him/her to be ignored, or to be told that you are not available (like
an answering machine)? You also need to tell Asterisk how to determine
if
the next "invite" comes
From: "Il Neofita" <[EMAIL PROTECTED]>
Date: Wed, 14 Feb 2007 20:37:52 -0500
This is the situation
A call me at my provider 1
I am not home and I would like to transfer the call
I do not pickup the call for some reason
I would like to hangup the caller, however, my asterisk try again to call
on
This is the situation
A call me at my provider 1
I am not home and I would like to transfer the call
I do not pickup the call for some reason
I would like to hangup the caller, however, my asterisk try again to call on
my mobile over and over
I would like to stop it.
Any idea?
Thank you a lot.
From: "Il Neofita" <[EMAIL PROTECTED]>
Date: Wed, 14 Feb 2007 19:30:51 -0500
I have this simple context
I am register to an external provider and when I am not home I would like
to transfer the phone outside
The problem that the call goes in loop
I don't see any loop in records below? What
I have this simple context
I am register to an external provider and when I am not home I would like to
transfer the phone outside
The problem that the call goes in loop
I cannot understand why.
Can you figure out my error?
Thank you
sip.conf
register => user:[EMAIL PROTECTED]/400
[inside]
Thomas Artner wrote:
> Hi!
>
> I am working hard on getting a useful attented transfer. (The built-in
> atxfer feature isnt useful - because of calls getting lost - has been
> discussed a few months ago)
>
> I have all my analog phones on sipura boxes. With the flash hook i can
> do such attended
Hi!
I am working hard on getting a useful attented transfer. (The built-in
atxfer feature isnt useful - because of calls getting lost - has been
discussed a few months ago)
I have all my analog phones on sipura boxes. With the flash hook i can
do such attended transfers without problems now.
But
Pablo Mora wrote:
[outgoing]
exten => 0,1,Dial,Zap/g1
exten => 0,2,Hangup
exten => 0,102,Congestion
You NEVER want Dial,Zap/g1
You If you want to just get an outside dialtone you ALWAYS want a trailing /
Dial,Zap/g1/
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chatt
Pablo Mora wrote:
>
Did you saw my dialplan? I don't think I would have to add r.
You never want to add "r" option to Dial()
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Change:
callprogress=yes
To:
callprogress=no
Also when dialing over the Zap FXO ports make sure to add the ww
before the DTMF digits so that your extension.conf reads like this:
exten => 9,1,Dial(Zap/g1/ww9 )
On 8/3/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
Ok
Here goes dialplan
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,
Can you please show your dialplan? with and withou the www?
Please only the dialplan that doesnt work. Also include
/etc/asterisk/zapata.conf
On 8/3/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
CF,
Adding www after Dial doesn't solve the trouble.
I think we are talking the same but I don'
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’
Yes, even with the r because the second it goes off hook it is
answered. This is an FXO port.
On 8/2/06, Jorge Mendoza <[EMAIL PROTECTED]> wrote:
Even if he has "r" in the dial plan?
Jorge
C F wrote:
> Then you have something wrong some other place, if you are using an
> FXO card then asterisk
Even if he has "r" in the dial plan?
Jorge
C F wrote:
> Then you have something wrong some other place, if you are using an
> FXO card then asterisk is not even giving you the ring, the panasonic
> is.
>
> On 8/2/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
>>
>>
>>
>>
>> I think still didn't explai
Then you have something wrong some other place, if you are using an
FXO card then asterisk is not even giving you the ring, the panasonic
is.
On 8/2/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
I think still didn't explain me clearly…
The problem is when I dial 0, in this case the asterisk t
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
You need to add a ww or 2 like this:
exten => 101,1,Dial(Zap/g1/ww${EXTEN})
or like this:
exten => 9,1,Dial(Zap/g1/ww9)
Hope this helps.
On 8/1/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
Ok,
I'm going to stop pictures
I have a Digium 4 FXO Card in my asterisk, and connect to Panasonic
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
Pablo, according to description I assume that you have an FXO at *
connected to an FXS port at Panasonic. If this is correct, could you
replace Asterisk by a telephone and see if it is possible to make call
to Ext1?
Jorge
Pablo Mora wrote:
> /Ok Ok, the figure doesn’t help./
> / /
> /Here we go a
Again you are not saying how asterisk is connected to the panasonic,
stop using pictures.
On 8/1/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
Ok Ok, the figure doesn't help.
Here we go again…
- -- --- --
| SIP | - | ASTERISK | -- | PA
Ok Ok, the figure doesn’t help. Here we go again… - -- --- --| SIP | - | ASTERISK | -- | PANASONIC | --- | PSTN | - -- --- -- | |
[mailto:[EMAIL PROTECTED]
Sent: Saturday, July 29, 2006
10:24 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Strange behaviour Panasonic KX-TD1232
Hello Pablo, I think you should decribe with details how are
you routing the call between the SIP
How is asterisk connected to the Panasonic KX-TD1232?
On 7/27/06, Pablo Mora <[EMAIL PROTECTED]> wrote:
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.
---
: [asterisk-users] Strange
behaviour Panasonic KX-TD1232
Hello,
Ive got asterisk running and
almost working with Panasonic KX-TD1232
I said almost, because theres a
strange behaviour when I make calls
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 | -
My guess is the problem is related to the reverse lookup of 147.135.20.128.
sip.broadvoice.com = 147.135.20.128
147.135.20.128 = non-existant
If you change your "host=sip.broadvoice.com" to "host=147.135.20.128"
it'll probably work until they change the IP of their server
Someone else who actua
Hi,
In sip.conf if
type=peer
then incoming calls from broadvoice are sent to context=incoming in
the extension.conf file.
But in sip.conf if
type=user
then we get the following messages in the packets being exchanged with
broadvoice:
1. "Found no matching peer or user for '147.135.20.128:5060'"
2
Problem:
> It seems the situation is improved when I remove the "regsiter => "
> statements in my sip.conf.
Cause:
If your internet connection is down, your DNS isn't working. IF your DNS
isn't working, it won't be able to resolve names. If it can't resolve a
name, it will sit there trying until
I have noticed a strange behaviour when our internet connection was
down a couple of hours last week.
What happens is that asterisk starts running *really* slow. If I type
"sip show peers" it sometimes responds correctly and shows all the
connected peers, but sometimes I get an empty list (this se
On Tue, 20 Jul 2004, Brian D'Arcy wrote:
> Anyone ever seen anything like this before?
Yes, with a Grandstream over an ADSL (routed) line. I disabled the check,
but the problem stil occured. And since this friends line was almost the
lowest ping in the complete ISP network something else must be
Hello all,
One of my remote employees is using a 7960 we sent him, on a public IP
address at his home office.
I've run pings and traceroutes both from the server to his phone, and
from the cable modem to our server, there's never a high ping time, or a
dropped packet, however about every 30 minut
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