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dahdi_maint -s 1
Span 1:
Framing Errors : 0:
CRC Errors : 0:
Code Violations : 247269:
E-bit Count : 0:
General Errored Seconds : 1998:
Any ideas what could be the problem?
Thanks!
Sebastian
Can you point me to the commit to see if i can backport it?
Thanks
El 07/05/2012 18:50, Jonathan Rose jr...@digium.com escribió:
- Original Message -
From: Sebastian Gutierrez scg...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Monday, May 7, 2012 10:38:03 AM
Subject
have you seen astmanproxy?
best regards
On Sep 23, 2014, at 10:05, jg webaccounts...@jgoettgens.de wrote:
Hi!
Maybe I have overlooked something, but I am sort of facing the following
problem. I always used the AMI interface to allow (older) client programs on
Windows to use their TAPI
the issue is with chan_sip not on rtp I will check wich commit break this
and fill an issue.
El mié., 5 de oct. de 2016 a la(s) 17:41, Sebastian <scg...@gmail.com>
escribió:
> From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop
> working, failing with
>
&g
>From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop
working, failing with
chan_sip.c:4083 retrans_pkt: Hanging up call
7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
the issue is fixed in current trunk head version
El jue., 6 de oct. de 2016 a la(s) 12:07, Sebastian <scg...@gmail.com>
escribió:
> the issue is with chan_sip not on rtp I will check wich commit break this
> and fill an issue.
>
>
> El mié., 5 de oct. de 2016 a la(s)
Maybe it could be accomplished in the firewall? Tell the firewall to NAT the
source port of packets to 5061?
Från: asterisk-users-boun...@lists.digium.com
För Alexander Perkins
Skickat: den 10 juli 2021 19:39
Till: asterisk-users@lists.digium.com
Ämne: [asterisk-users] SIP Source Port
Hi
)
exten = 2122020683,2,Hangup
--
thank you very much
sebastian
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Hi,
how can I completely disable silence suppresion and echo cancelling in
asterisk (and zaphfc)
Thank you very much.
Sebastian
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=_01186.,1,dial(SIP/[EMAIL PROTECTED],30)
exten=_011886.,1,dial(SIP/[EMAIL PROTECTED],30)
exten=_011972.,1,dial(SIP/[EMAIL PROTECTED],30)
exten=_011.,2,congestion() ; No answer, nothing
exten=_011.,102,busy() ; Busy
Thank you very much
Sebastian
are of very good quality)
(only works from 2000 to 2019 (-:, which should be enough for the moment )
I adjusted some sleeps beween the digits to make it sound more natural.
feel free to send me your suggestions (it's my first agi script ever),
PS: is there a sound file which contains the ?
/sebastian
Dear Asterisk Users,
if I do a : /usr/sbin/asterisk -r -x restart gracefully , asterisk
just quits without any message. Any idea ?
(debian 3.1 with asterisk packages from unstable :
1.0.7-BRIstuffed-0.2.0-RC7k)
/sebastian
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Can you send me the patch?
SA
-Mensaje original-
De: Geoff Speicher [mailto:[EMAIL PROTECTED]
Enviado el: Sábado, 29 de Enero de 2005 23:11
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]
Sipura has implemented auto-answer in
Someone know what kind of terminal I need to use for this feature?
What exactly do this and what is way to use that?
Sebastián Atala
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Voipproviderlist.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jeromie Reeves
Enviado el: Martes, 04 de Enero de 2005 03:30 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] OT: List of VoIP providers?
I
=0.8
language=de
[interfaces]
msn=4132
incomingmsn=*
controller=1
context=demo
mode=immediate
isdnmode=ptp
devices=30
extensions.conf:
[demo]
exten = 4132,6,Dial(SIP/test)
so, I'm a bit confused now.
what can I do???
thanks for helping me out!
greetings, Sebastian
Capi Debug output:
CAPI
I can send a list, mobile is not complete but it has a lot of numbers...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de PHP Mechanic
Enviado el: Viernes, 07 de Enero de 2005 11:57 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto:
Try with ASTCC is free.
Sebastian
-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Enviado el: Martes, 14 de Enero de 2003 14:56
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] PrePaid Applications
Hi;
Is the Prepaid Applications that we can use
Here is the link
http://www.voip-info.org/wiki-ASTCC
SA
-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Enviado el: Martes, 14 de Enero de 2003 18:21
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] ASTCC
Dear Sebastian;
Thanks
Can Asterisk only send and receive SIP packet without media proxy in any
time? I am using re-invite but I don't want that the ring back is proxy by
asterisk.
Someone knows a way to do that?
Sebastian
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http://en.wikipedia.org/wiki/HDLC
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Eric Bishop
Gesendet: Montag, 31. Januar 2005 11:40
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [Asterisk-Users] HDLC for Dummies?
Can
Which version of Asterisk this did work?
Sebastián Atala
-Mensaje original-
De: Geoff Speicher [mailto:[EMAIL PROTECTED]
Enviado el: Sábado, 29 de Enero de 2005 23:11
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Sipura SPA-841 auto-answer support [patch]
Hello everybody,
Anyone knows where I can find information for configure the Asterisk as
MGCP-H323 transcoder?
May be an example or something.
Thank you very much
Best regards
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that their call is about to be
answered.
Any ideas?
Regards,
Sebastian.
--
Sebastian Filzek
Teragen International Pty Ltd
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hello all, I am
having a trouble with Audio using h.323 channel...
I am doing
this
Call comes into
cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and
send call to a SoftSwitch that routes the call, I can see log debug telling me,
CALLED XXX, and then RINGING,
ldconfig, check that /etc/ld.so.conf have path to where
oh323 library is
and then run ldconfig
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Fathallah
SoumayaEnviado el: Martes, 13 de Julio de 2004 12:27
p.m.Para: [EMAIL PROTECTED]Asunto: Re:
[Asterisk-Users]
IN MY HONEST OPINION... IMHO
I am right?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de ruixun wu
Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] How to uninstall Asterisk?
hi Gus and Roger,
I tried a lot of
times to get it worked, but I cant obtain audio using SIP-chan_h323 or
chan_h323-SIP
I tried disbling
FastStart without good results...
What's the
problem?
I need to do BRIDGE
between SIP and H.323!!
help!!
Sebastian.-
it's also In My Humble Opinion too.
Gonzalo
P/D: Como andas Seba... :)
On Wed, 2004-07-14 at 11:45 -0300, Sebastian Nocetti wrote:
IN MY HONEST OPINION... IMHO
I am right?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de ruixun wu
Enviado el
Hello, I want to
know what kind of equipment I need to handle 120 simultaneous calls with a
Digium 4E1 card... and using 120 G.729 licences some
help?
thanks
Sebastian.
I cant do SIP -
CHAN_H323 transmit audio!!! I can hear rings, but when connected,
NOTHING
It happened in both:
SIP - CHAN_H323 and CHAN_H323 - SIP...
when it will be
solved?
a.m.Para: [EMAIL PROTECTED]Asunto: RE:
[Asterisk-Users] STILL NO AUDIO
Happen to have any NAT
in the mix?
bkw
-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian NocettiSent: Monday, July 19, 2004 9:25
AMTo:
[EMAIL PROTECTED]Subject
put disallow=all and allow=ulaw in
sip.conf and h323.conf (and NO OTHER ALLOW= LINES)
On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote:
I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when
connected, NOTHING
It happened in both: SIP - CHAN_H323 and CHAN_H323
Testing both...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Michael Manousos
Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] STILL NO AUDIO
Why don't you use asterisk-oh323?
Michael.
Sebastian
What kind of problem?
All works OK except that config
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Holger Schurig
Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] STILL NO AUDIO
I WANT TO USE
To translate with g729 you need licenses...
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Brent Franks
Enviado el: Martes, 20 de Julio de 2004 10:01 a.m.
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] codec translate
HI ALL;
Is astersik
Read README in oh323 directory, use exactly libraries you can read there,
and obviusly apply patch first...
Then run ldconfig
Put variables on environment
And all is ok
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de ruixun wu
Enviado el: Viernes, 23 de
does not log out.
Does anyone know what the data tacked on the end of the SIP name is
and how to stop it?
Regards,
Sab.
--
Sebastian Filzek
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Hello all, somebody
can tell me how h.323 status is? it is working OK?... it has implemented
faststart and tunneling per peer based?...
thanks a
lot!!
Sebastian from
Argentina.
---
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.791 / Virus Database: 535
I want to configure the voicemail, extension, agent, queue and sip from
postgres. Someone have experience in that?
Someone know how can I configure meetme without a Zaptel card?
Sebastián
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and install it without
error. I was looking for solution in list and internet but i dont find
anything.
What can be a problem ?
Sebastian
Bojczuk
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Check what IOS ata have installed... Because by default it does not comes
with H.323 - SIP IOS...
If you want I can send you both ios...
Contact me at: [EMAIL PROTECTED]
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Rodney Acosta
Coya
Enviado el:
Is anyone successfully using asterisk-prepaid-0.3.1?
I try to configure but doesn't work. It said that you need to do a few step,
copy a few files and that is.
Please, if someone has any tips about the configuration, answer me.
Sebastian
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: Asterisk Users Mailing List - Non-Commercial Discussion
CC: Sebastian Atala
Asunto: Re: [Asterisk-Users] Prepaid
I use ASTCC and works perfect for Prepaid situations.
Nhauel Ramos.
On Mon, 29 Nov 2004 16:48:46 -0400, Sebastian Atala [EMAIL PROTECTED]
wrote:
Is anyone successfully using
I think you CAN'T DO VOIP-VOIP into CISCO Equipment, it have to be POTS-VOIP
or viceversa.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Brian Wilkins
Enviado el: Martes, 30 de Noviembre de 2004 05:57 a.m.
Para: Asterisk Users Mailing List -
(Netherlands)
zaphfc: card 0 layer 1 state = G2
zaphfc: card 0 layer 1 state = G3
---
Thank you in advance !
Sebastian
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not initialize ..., due to , this had helped alot and had
saved alot of hours (-;
/sebastian
Pascal C. Kocher wrote:
Hello
Make sure you run ztcfg only once(!) per reboot. A second time seems to
kill the zaphfc module (even if it doesn't state an error)
Do you have any wcfx* cards running?
Best regards
I am doing that actually, terminating calls via SIP on a
Cisco AS5300, and it is working good.
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 10:43
a.m.Para: [EMAIL PROTECTED]Asunto:
[Asterisk-Users] Asterisk + AS5300
de 2004 11:05
a.m.Para: Asterisk Users Mailing List - Non-Commercial
DiscussionAsunto: Re: [Asterisk-Users] Asterisk +
AS5300
Can you post a sample of your configuration?
(sip.conf, extensions.conf and as5300 dial-peers)
Thanks!
boch.-
- Original Message -
From:
Sebastian
help you more if you do.
/sebastian
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On 16.12.2004 Martin List-Petersen Wrote
[EMAIL PROTECTED]:
then the routing to SIP-Phones shall be based on the MSN-Configuration.
means, if someone dials 4321-1000 the call shall go to SIP/boss
and 4321-1001 to SIP/secretary
and so on.
is this just by adding an
exten =
)
to the context set in the /etc/asterisk/capi.conf?
and what to do, so that, if the boss calls out the MSN of the secretary
is shown?
and if the secretary calls out also their MSN is shown?
thank you for helping!
Sebastian
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[EMAIL
is h323 per user
based working??? I have setup this:
[User1]type=userhost=xx.xx.xx.xx
context=international
incominglimit=30
But all calls from
xx.xx.xx.xx are not routed to context international, it is
working?
I am using
chan_h323
Thanks!!
Sebastian
Nocetti.
---
Checked
Thanks !! I will try!!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Soren Rathje
Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] h.323 Type=User
Sebastian
Sebastian Nocetti wrote:
is h323 per user based working??? I have setup this:
[User1]
type=user
host=xx.xx.xx.xx
context=international
incominglimit=30
But all calls from xx.xx.xx.xx are not routed to context
international, it is working?
I am using chan_h323
I'm using current CVS 21
I am interested too in termination using SIP to brazil, we need h.323 too...
Can you contact me?
Thanks
Sebastian.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Daniel Bichara
Enviado el: Martes, 21 de Septiembre de 2004 11:06 a.m.
Para: [EMAIL
Asterisk works ok, but it have a lot of errors...
1st: It ever handle audio packet, and you cant do for exacmple only
SIGNALLING
2st: It cant handle more than 20 channels simultaneous ... I tested it.
3st: It does not have fully Radius support.-
-Mensaje original-
De: [EMAIL PROTECTED]
by myself
Thanks in Advance,
and excuse my bad English ;)
Sebastian Mauer
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Title: Mensaje
Hi, I have this
scenario
Cisco 5300 (public
ip. 200.47.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- Cisco
3600 (public ip: 64.76.xx.xx , same network than * )
When a calls comes
in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome
message and
;
charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
MensajeFijate en los 'voice codecs' de los dial-peers.
- Original Message -=20
From: Sebastian Nocetti=20
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 12:41 PM
Subject: [Asterisk-Users] Media Negotiation
Does Asterisk support Radius accounting?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
[EMAIL PROTECTED]
Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m.
Para: [EMAIL PROTECTED]
Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs
Send
PROTECTED]
Subject: Re: [Asterisk-Users] Radius on *
Reply-To: [EMAIL PROTECTED]
Sebastian Nocetti wrote:
Does Asterisk support Radius accounting?
No and there is absolutely no need for it to. RADIUS is not anything
that should have ever been deployed in a VoIP environment
Title: Mensaje
Hi, I want to use
G.723.1 on *, I read it is supported in Pass Through mode, but I don't
understand whats the meaning of that.
I have a GW 5300 and
an ATA 186 and I want to place calls to PSTN.
I setup this
config:
[general]port =
5060
bindaddr =
xx.xx.xx.xx
context =
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing source?
kind regards
Sebastian
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the wcte12xp driver,
because of an other card.
Anybody knows what my problem is?
kind regards
Sebastian
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)
Channel 31: Clear channel (Default) (Slaves: 31)
31 channels configured.
kind regards
Sebastian
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Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com wrote:
Sebastian Reitenbach wrote:
Hi,
When you type ztcfg -vvv, what does it display?
How about your zaptel.conf, zapata.conf and the snip of your dial plan
it in the first place.
I added a rmmod wcte11xp modprobe wcte11xp to /etc/init.d/zaptel into the
start section, and when I then login I have no problem to start asterisk.
Sebastian
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asterisk
-06-14 07:25 9
crw-rw 1 asterisk asterisk 196, 254 2007-06-14 07:25 channel
crw-rw 1 asterisk asterisk 196, 0 2007-06-14 07:25 ctl
crw-rw 1 asterisk asterisk 196, 255 2007-06-14 07:25 pseudo
crw-rw 1 asterisk asterisk 196, 253 2007-06-14 07:25 timer
kind regards
Sebastian
]: ast_h323.cxx:169 void PAssertFunc(const
char*): Assertion fail: Invalid parameter, file ../common/sockets.cxx,
line 1354, Error=115
what is the problem?
Sebastian BOZIOREANU
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Hi,
I'm trying to make odbcexec work with Asterisk 1.6.
I had the attached code (app_odbcexec, not the standard one) working great
with asterisk 1.2 an MSSQL Server on heavy load PBXs with no problem, I'm
trying to port this to asterisk 1.6 but I'm failing to do so.
I attach de working code in
-Commercial Discussion
Asunto: Re: [asterisk-users] ODBCExec from Dialplan
On Thu, 2008-11-06 at 12:16 -0200, Sebastian Gutierrez wrote:
I'm trying to make odbcexec work with Asterisk 1.6.
Why not just use the functionality of func_odbc already built into
Asterisk 1.6? Is there something you gain
Ok, sorry for the response on the same thread.
This is a new one.
The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network
Discussion
Asunto: Re: [asterisk-users] ODBCExec and Asterisk 1.6 New Thread
On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote:
The main thing is that with this I set the Store Procedure or Query
directly on the dialplan line, is easier to configure, change, manage,
etc.
I also know
and Asterisk 1.6 New Thread
On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
Dou you have any example? Can I call directly to querys without the
templates???
func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something
extensions.conf:
exten = 123,1,Set(result=${ODBC_EXEC(SELECT foo
Anyone is using 1.6 in production??
Is it ready?
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Production ready??
Steve Totaro wrote:
On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Sebastian Gutierrez wrote:
Anyone is using 1.6 in production??
Is it ready?
I have a number of people using 1.6
Hi,
How can I pass the following data to te queuelog via ami??
Agent,data.
??
I'm doing this:
Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n
And thath works fine getting the log with the event but I cant find how to
pass the agent and data parameters
Any
-Commercial Discussion
Asunto: Re: [asterisk-users] QueueLog from AMI
On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
Hi,
How can I pass the following data to te queuelog via ami??
Agent,data.
??
I'm doing this:
Action: QueueLog\r\nQueue: queueprueba
] QueueLog from AMI
On Wed, Nov 12, 2008 at 7:31 PM, Sebastian Gutierrez
[EMAIL PROTECTED] wrote:
Not if I have realtime, I'm inserting and deleting from queue_members
table,
so I don't have that info.
As am I.
I posted a patch that fixes this, so you could be interested in
keeping it in mind
mailbox.
Can anybody helpme with that please?
Thanks very much in advance
Sebastian
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Are my e-mails arriving to the list? can somebody confirm?
Sebastian
-- Forwarded message --
Date: Fri, Nov 21, 2008 at 11:06 AM
Subject: SPA2100 transfer to ASTERISK CID
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi all,
I
with another device with an
in-box E1?
Thanks very much in advance,
Sebastian
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?, or just configure the Alcatel PBX to work
with default TEXXXP card configuration?.
Is there any issues with MFC-R2 and Asterisk cards?
Best Regards,
Sebastian
On Fri, Apr 17, 2009 at 4:48 PM, Jean-Denis Girard jd.gir...@sysnux.pfwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Sebastian
about astbill, except a few lines y Wiki. Is there
something like that?
Thanks very much in advance,
Sebastian
e-mail:[EMAIL PROTECTED]
msn:[EMAIL PROTECTED]
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Hi,
I'm having a problem with chanspy.
I have configured this way
Chanspy(SIP/1)
So it scans all my 1XXX extensions.
That's working just fine, but when I try to switch to an extension ej. 1234#
(it has a call in progress), but the chanspy jumps to another extension, no
te one I selected.
Hi,
I'm having a problem with chanspy.
I have configured this way
Chanspy(SIP/1)
So it scans all my 1XXX extensions.
That's working just fine, but when I try to switch to an extension ej. 1234#
(it has a call in progress), but the chanspy jumps to another extension, no
te one I selected.
the feeling that I am not the first one with that
problem, so I want to ask for more easily/robust tests to make sure the master
is running or not.
any suggestions are appreciated.
kind regards
Sebastian
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portugese.
Sebastian
Leonardo Silva [EMAIL PROTECTED] wrote:
Hi Sebastian,
This url http://underlinux.com.br/content/view/6330/70/ have some thinks
that you need.
Leonardo Silva
2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]:
Hi,
I have setup two asterisks
the 'catchallvoip' trunk. Does this sound reasonable or can the Goto cmd not be used to switch contexts for outbound calling rules by extensions?Thanks in advance for any insights!-Sebastian
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http
on the username? Does asterisk allows two
extension sections with the same number?:
[2000]
username=companyA_2000
context=contextCompanyA
[2000]
username=companyB_2000
context=contextCompanyB
Any help will be appreciated.
Sebas
--
Sebastian Silva
G R U P O G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
: Comp-A user 2000 calls comp-B user 2000 by dialing 72000.
--
Sebastian Silva
G R U P O G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706- (int. 121)
[EMAIL PROTECTED]
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Perfect, that's exactly what I need.
I will try that, thanks a lot.
Sebas
Matt Riddell wrote:
Sebastian Silva wrote:
Hi everybody,
I am writing here because I can't find the solution to my problem (my
asterisk configuration). I hope somebody can give me a hand with it:
I need to provide a PBX
Hi,
Someone knows how can I register my Asterisk to a gatekeeper using
zone parameters?
I'm using asterisk 1.0.7 and oh323 0.6.5.
I'm trying to register to a gatekeeper in another network and I can't reach
this with a broadcast.
Zone is the name who Cisco call the GK identification.
a backtrace. The machine I´m using for asterisk is
running fli4l which has no gdb available.
Maybe somebody could give me a hint looking at the attatched logs? Cards are
Fritz!DSL (with integrated isdn port) and Acer isdn surf 128.
Thank you,
Sebastian
8
about.
thanks a lot.
Sebas
--
Sebastian Silva
G R U P O G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706- (int. 121)
[EMAIL PROTECTED]
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the caller typed 123114.
I can live with fixed length extensions. means, always wait for 3
digits.
thanks for help..
Sebastian
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