Hello all,
I have the following challenge: I have to add a variable to the
destination channel with the following conditions:
1) It has to be set in the dialplan, in runtime.
2) The source channel can't have the same variable has the destination.
I had two ideas so far, but they seem
wrote:
Alexandre Rodrigues wrote:
Hello all,
Hola,
I have the following challenge: I have to add a variable to the
destination channel with the following conditions:
1) It has to be set in the dialplan, in runtime.
2) The source channel can't have the same variable has
Hello Marcus,
Had the same problem, looked in the internet and found your question.
Since I now have an answer
I will put it here! :)
In the dialplan I had this:
exten = 4000,1,Dial(Dahdi/4)
exten = 4000,n,hangup()
exten = _4,1,Dial(IAX2/PBX/${EXTEN:1})
exten =
Hello,
Facing the same problem with the following debug skinny log:
-- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23
Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S :
Invalid SCCP message! : ID :83
Received Alarm Message: 32: Name=SEP001X
Solve it. :)
Found this link: http://www.voip-info.org/wiki/view/SCCP-HOWTO2
Cheers,
Alex
2012/3/22 Alexandre Rodrigues alex...@gmail.com:
Hello,
Facing the same problem with the following debug skinny log:
-- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23
Hello all,
In my PBX, AMI does not generate Hold or MoH events for a IAX trunk.
I did the following tests with different channels:
Dahdi to SIP: Hold and MoH events.
Dahdi to Dahdi: MoH events.
SIP to SIP:Hold and MoH events.
IAX to SIP:Hold events.
...@xorcom.com:
On Mon, Dec 21, 2009 at 04:19:46PM +, Alexandre Rodrigues wrote:
Hello all,
I am still studing the problem and I'm now focusing on the Disconnect
supervision asterisk issue.
I which to obtain some feedback of the ideas I had to resolve this problem.
I will set busydetect
Hello Carlos,
I have the same problem when I try to do a flash hook with dahdi module.
Did you resolve your problem?
Thanks in advance,
Alex
2009/8/13 Carlos Rojas crt.ro...@gmail.com:
Hello everybody
I have an asterisk with an integration of alcatel pbx, by sip trunk, all
calls are fine,
Hello Giedrius,
Same problem with me in asterisk 1.4 using a dahdi module. I am trying
to transfer a call using
flash hook but I get the same message.
Did you resolved this issue?
Thanks in advance,
Alex
2009/8/18 Giedrius Augys voi...@gmail.com:
Hello,
I've a problem. I've asterisk
Hello all,
I want to park calls using the callpark application, but I don't want to
hear the saydigit when the called is parked.
To resolve this issue I use the following instruction in the dialplan:
exten = _8XX,1,ParkAndAnnounce(|1000|local/1...@default|)
Because local/1...@default is
Of *Alexandre
Rodrigues
*Subject:* [asterisk-users] parkcall: How to remove announcement.
Hello all,
I want to park calls using the callpark application, but I don't want to
hear the saydigit when the called is parked.
To resolve this issue I use the following instruction in the dialplan
Thanks very much for your help! :)
2010/8/16 Danny Nicholas da...@debsinc.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Subject: Re: [asterisk-users] parkcall: How to remove announcement.
Hi!
How can
Hello all,
My pbx server is connected to a sip gateway, when I call an originate
command from the asterisk console, to establish a sip connection, the
gateway doesn't accept URL with white spaces, for example:
* Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport *
* From:
.
(Closes issue #13810)
/
can this happen because of the sequence number??
Thanks in advance,
Alex
2010/5/5 Alexandre Rodrigues alex...@gmail.com
Hi all,
I am trying
Hi all,
I am trying to connect to a softphone application using an Iax channel on
Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but
not inbound from asterisk to softphone.
I get the following Debug:
--
Hello asterisk users,
I am having quite a problem finding, in the misdn.conf file, the accountcode
variable.
In sip and dahdi the variable name is account code, is there any kind of
variable to set this property in misdn.
Thanks in advance,
Alex
--
I am having problems with spa3102 FXO ports. It has a lot of echo, so be
careful when you get one of does!
2010/1/6 Joseph L. Casale jcas...@activenetwerx.com
I don't use them myself, but I was thinking that the RHEL5 spec files
might be another place to look for what you need to build with
Hello all,
I have a Linksys spa3102 with one FXS and one FXO port.
The problem is that I have a lot of echo when using the fxo port, the sound
is of very low quality.
So, since I am passing from a FXO port to a SIP channel I ask:
is there any Sip echo canceler software for
Problem solved. :)
I was adding to the pstn line a gain of 6 DB for both sides.
It has to be less than zero. After that the echo almost disappeared.
2010/1/21 Alexandre Rodrigues alex...@gmail.com
Hello all,
I have a Linksys spa3102 with one FXS and one FXO port.
The problem is that I
in advance,
Alex.
2009/12/17 Alexandre Rodrigues alex...@gmail.com
Yes, It happens also, I am having this problem in many servers with
different boards, asterisk versions and drivers (zap and dahdi). Sometimes
in one month it is all fine in the other I get a lot of this off hook
states
a good analogue FXS card without the off hook state
problem???
Thanks in advance.
Alex
2009/12/4 Alexandre Rodrigues alex...@gmail.com
Hello again,
Adding more information:
Core show channels:
Channel Location State Application(Data)
DAHDI/4-1
electrical problem that happens occasionally that
trigers this jammed state??
Thanks
Alex
2009/12/17 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Dec 04, 2009 at 03:33:00PM +, Alexandre Rodrigues wrote:
Hello again,
Adding more information:
Core show channels:
Channel
Of *Alexandre
Rodrigues
*Sent:* Thursday, December 17, 2009 12:02 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Rsrvd state and off hook dahdi issue
Hello Tzafrir Cohen,
First of all thanks for the reply. :)
The fax normally works. But two
Hello again,
Adding more information:
Core show channels:
Channel Location State Application(Data)
DAHDI/4-1s...@national_mobile:1 Rsrvd(None)
DAHDI/1-1s...@national_mobile:1 Rsrvd(None)
Dahdi show channels:
Hello all,
I have asterisk 1.4.26.2 working on linux CentOS 2.6.18 with a dahdi
driver 2.2.0.2 and Digium Wildcard AEX410.
I configured 3 ports of 4 all fxs. The port 1 is connected to a
analogue phone, the port 3 and 4 are connected to two analogue faxes.
dahdi-channels.conf:
;;;
that auto-answers and hangs
up, so no agent action is needed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandre
Rodrigues
Sent: Friday, October 30, 2009 8:24 AM
To: asterisk-users@lists.digium.com
hello all,
I have asterisk 1.4.26 working on CentOS 5.3 and I have the following problem:
- when I restart asterisk all the members of the queue are Invalid.
- when I make a call to one of the members, of the queue, and then
check the state, it turns to Not in use for the called phone,
Hello all,
I am quite new in asterisk and I am trying to create a dialplan that
executes the following steps:
1. A SIP friend dials 102 extension.
2. Asterisk PBX responds with some beeps.
3. The sip friend hangs up the phone.
4. Asterisk PBX calls back to the sip friend after 30 seconds
28 matches
Mail list logo
