I am unable how to get a zomebie sip channel to hangup. I've tried the
following in the manager but it doesn't work.
Action: Status
Response: Success
Message: Channel status will follow
Event: Status
Privilege: Call
Channel: SIP/2003-09e2bbe8ZOMBIE
CallerID: 093611168
CallerIDName: unknown
Benny Amorsen wrote:
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA If it's not a security issue I might as well have all phones with
LA context=default in sip.conf even though voip-info specifically
LA warns against that. Wonder why?
Random SIP calls from the internet could end up in
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 22 Feb 2007, at 07:25, Tzafrir Cohen wrote:
I tried to use the 1.2.x RPMs and they would not work for me
attempting to use them with an Eicon Diva Server card and Melware's
chan_capi. Only by looking at the SRPM did I notice that they are
Hi
the new asterisk 1.4 supports to store queue log information directly
into a database? (like CDR) ?
thanks
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
I have an Asterisk 1.2.9.1 with chan_misdn 0.3.1-rc23 and a beronet
octoBRI on a Debian box I have to set up instead of an old legacy PBX.
My problem is I get only the base DNID and not the extensions (the last
two digits) in Asterisk but the old PBX got all the DNID number so I
think it
hi,
is there anyway to Answer() the caller channel after the called number
pickedup the phone.
when an outside caller calls * system just continue ringing and not pick up
the line and just dial an extension and then answer the caller channel after
the called extension picked up the phone.
is this
context 'default' has not any special, it's context, that will be used
if your peers/users definition doesn't contain any specific context
if you have permited 'anonymous' calls to your asterisk, i.e
allowguest=yes, unautenticated calls (calls, that will not match any
specific user in sip.conf)
Hi all,
I have read many forums and discussion groups talking about fax
support in asterisk. Some of them conclude that asterisk doesn't
support fax. However, some of them conclude that there is no
relationship between fax and asterisk as asterisk will only pass the
fax signal to the fax
I think, this can be solved using phone autoanswer feature, look at wiki...
exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten = s,2,Dial(SIP/myphone)
Paradise Dove wrote:
hi,
is there anyway to Answer() the caller channel after the called number
pickedup the phone.
when
On Thu, 22 Feb 2007, Rilawich Ango wrote:
Hi all,
I have read many forums and discussion groups talking about fax
support in asterisk. Some of them conclude that asterisk doesn't
support fax. However, some of them conclude that there is no
relationship between fax and asterisk as asterisk
Hello ango,
Try asterfax, I use it and it works fine.
If you want to use a regular fax machine you can use an iaxt device to
connect your fax machine to the LAN.
Regards,
mjmarrio
On Thu, 2007-02-22 at 16:32 +0800, Rilawich Ango wrote:
Hi all,
I have read many forums and discussion
But, as shipped, asterisk doesn't have native fax support,
but it can be
patched in via the spandsp code, giving you 2 new
applications: RxFax and
TxFax. You can plumb an incoming answered call to RxFax and
it will decode
the incoming fax stream into a TIFF file which you can then
From: Larry Alkoff [EMAIL PROTECTED]
Date: Wed, 21 Feb 2007 20:00:52 -0600
...
You should consider that if any channel, incoming line, etc can enter an
extension context that it has the capability of accessing any extension
within that context.
Therefore, you should NOT allow access to
21 feb 2007 kl. 15.50 skrev James Fromm:
Anybody seen this behavior?
To determine if it's my config or a bug, could I trouble someone
running Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP
interface as a test? After a few hours a 'sip show inuse' should
indicate the
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command? Thanks
for pointers.
None.
What do you want to do with SIP INFO?
/O
___
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asterisk-users
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look at wiki...
exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten = s,2,Dial(SIP/myphone)
Or without. One of my contexts is set up exactly
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
Look at:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
These messages also
22 feb 2007 kl. 10.05 skrev Gordon Henderson:
On Thu, 22 Feb 2007, Rilawich Ango wrote:
Hi all,
I have read many forums and discussion groups talking about fax
support in asterisk. Some of them conclude that asterisk doesn't
support fax. However, some of them conclude that there is no
OEJ == Olle E Johansson [EMAIL PROTECTED] writes:
OEJ And for fax over VOIP, sometimes called FOIP, Asterisk 1.4.x
OEJ supports T.38 passthrough. However, the 1.4.0 release is buggy,
OEJ so either use 1.4 from subversion or wait for 1.4.1.
T.38 passthrough is not very exciting unless you happen
Well, but isn't lines that begin with -- on the same verbosity level?
So lowering the verbosity would in this case mean that you also stop
displaying the dialplan execution steps. I have a similar problem
regarding the -- SIP Seeding peer from astdb messages. I get a lot of
these, so I tried
genzaptelconf you mean?
2007/2/22, Paul Hales [EMAIL PROTECTED]:
genzaptel is _not_ your friend when setting up E1.
PaulH
On Thu, 2007-02-22 at 00:46 +, younss azzayani wrote:
this is my zaptel.conf::
[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
# Autogenerated by
22 feb 2007 kl. 11.19 skrev Torbjörn Abrahamsson:
Well, but isn't lines that begin with -- on the same verbosity
level? So lowering the verbosity would in this case mean that you
also stop displaying the dialplan execution steps. I have a similar
problem regarding the -- SIP Seeding peer
when i m running genzaptelconf
[EMAIL PROTECTED] ~]# genzaptelconf
STOPPING ASTERISK
STOPPING FOP SERVER
safe_opserver: no process killed
FOP Server Stopped
Generating '/etc/zaptel.conf'
Generating '/etc/asterisk/zapata-auto.conf'
Unloading zaptel hardware drivers:
Unloading rxt1: ERROR:
We have recently purchased a B410P Digium 4* ISDN-2 card with hardware EC.
On the same server, I also have a regular Digium 4-channel PSTN-card
(TDM410P ?), used to interface to some analog devices, a.o. 2 fax machines.
For faxing, EC needs to be off (or so I understand from the archives).
I do need MWI notifcation, just not on this particulary trunk. Is there
anyway to to turn off MWI on a particular trunk or can it only be done
globally?
On 2/22/07, Olle E Johansson [EMAIL PROTECTED] wrote:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to
21 feb 2007 kl. 12.54 skrev Steve Langstaff:
Hi All.
This is on Asterisk 1.2.13
I place a call between 2 SIP phones (with canreinvite=yes,
qualify=yes).
I reset the phones (so they don't have time to say BYE).
Asterisk seems to think that the call is still ongoing. This persists
until I
Agreed, but your response to the OP said to lower the verbosity, and I
commented that it might not be possible, due to then seeing no dialplan
execution... :)
How about the seeding messages then? Will you move these to a debug
level? Or do a bug need to be filed in Mantis?
// Tobbe
Olle E
Are the RTP timers applicable with canreinvite=yes ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Olle E Johansson
Sent: 22 February 2007 10:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Channels
Hello,
I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX
calling group. This allows me to have 4 concurrent calls.
I want to use this connection with my VoIP network, with an Asterisk PBX, so I
need a ISPBX to Voip (SIP) gateway. The problem is that I can't find any
Hi,
This is my first post to the list so please be gentle ;)
Okay, I have successfully configured Asterisk with a X100P clone card (soon to
be replaced with a 1xFXO,1xFXS TDM card), and it quite happily answers the PSTN
line and routes it to either a extension or voicemail.
What I would like
Hi,
I am trying to branch a call based on it's duration but
${CDR(duration)} is always 0.
(The idea is to keep ringing the operator until a certain amount of
time has lapsed)
This does not work:
exten = s,4,Background(local/script8)
exten = s,5,Dial(${OPERATOR},30,tr)
exten =
Marnus van Niekerk wrote:
Hi,
I am trying to branch a call based on it's duration but
${CDR(duration)} is always 0.
(The idea is to keep ringing the operator until a certain amount of
time has lapsed)
This does not work:
exten = s,5,Dial(${OPERATOR},30,tr)
Change the 30 on your dial
On Thu, Feb 22, 2007 at 09:29:47AM +0200, Tzafrir Cohen wrote:
On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote:
bristuff is the only patch in functionality, and for 1.2.15 I need to
drop it again, because it does not apply
Gee, it shows you're not on the bristuff list.
Hi
Sorry if this comes twice; i sendt first version from non-member address.
I'm learning use Asterisk but cannot solve following problem: i have
Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to
register within LAN i got 'Forbidden' message from Linphonec.
22 feb 2007 kl. 11.36 skrev Eric Bishop:
I do need MWI notifcation, just not on this particulary trunk. Is
there anyway to to turn off MWI on a particular trunk or can it
only be done globally?
You enable it per device in sip.conf - that's the only way.
/O
22 feb 2007 kl. 11.54 skrev Torbjörn Abrahamsson:
Agreed, but your response to the OP said to lower the verbosity,
and I commented that it might not be possible, due to then seeing
no dialplan execution... :)
Well if you want that level of detail during the execution, these
error messages
22 feb 2007 kl. 12.20 skrev Steve Langstaff:
Are the RTP timers applicable with canreinvite=yes ?
how could we possibly check RTP if the RTP doesn't touch or network
card at all?
The timers are only used when we have RTP streams going to us. If the
RTP stream
is redirected, it's up to
Hi guys,
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
What it mean ?
Thanks.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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On Thu, Feb 22, 2007 at 01:11:03PM +0100, Axel Thimm wrote:
On Thu, Feb 22, 2007 at 09:29:47AM +0200, Tzafrir Cohen wrote:
On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote:
bristuff is the only patch in functionality, and for 1.2.15 I need to
drop it again, because it does not
Apologies
On Thu, 22 Feb 2007 11:49:52 +, uxbod [EMAIL PROTECTED] wrote:
Hi,
This is my first post to the list so please be gentle ;)
Okay, I have successfully configured Asterisk with a X100P clone card
(soon to be replaced with a 1xFXO,1xFXS TDM card), and it quite happily
answers
OK, nice. Any chance of it finding its way into 1.2-branch?
I agree in some extent that one get a lot of information when looking at
the dialplan execution, but the difference is that this is usefull
information. Looking at the dialplan pass by is not made easier by
having the seeding
Hello,
I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and
Nortel TX-1. I had problems with name transfer and with the help of Cisco
support I've fixed it. Enclosed here are the definitions needed for it.
BTW, Cisco's CCM is using MGCP thus the Q.sig is handled by CCM.
22 February 2007 12:22, Olle E Johansson wrote:
22 feb 2007 kl. 12.20 skrev Steve Langstaff:
Are the RTP timers applicable with canreinvite=yes ?
how could we possibly check RTP if the RTP doesn't touch or
network card at all?
You can't. I realise.
The timers are only used when we
From: Jason Aarons \(US\) [EMAIL PROTECTED]
Glad to hear you had a workaround.
I would suggest re-queing your TAC case, perhaps you got a outsourced or
less experienced engineer at Cisco. Their support has varied depending on
which city/group you get. Some have more experience then
I have been given the task of getting a Polycom IP 601 running SIP
v2.0.1.0291 to register with our SER proxy and be able to interact with
our Asterisk server for voice mail. The Asterisk server currently sends
unsolicited NOTIFY messages to turn on/off the message waiting light.
Most of
Try to see maybe it could be done with Patton Smartnode series gateway.
Hector Rivas Gandara wrote:
Hello,
I have 2 ISDN BRI connections, configured by my telephony provider as a ISPBX
calling group. This allows me to have 4 concurrent calls.
I want to use this connection with my VoIP
On 02/22/07 06:04 Michelle Dupuis said the following:
I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster). The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
I have attached the SIP debug output below.
Hi,
I managed to get SIP re-invite working. If a call comes into my * box
from my ITSP on a DiD, I can handle the call by calling Dial() in my dial
plan and the call will get transferred and the media does not pass through
my * box after the call is bridged.
However, if I Answer() the call
Try http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways
- Original Message -
From: Mindaugas Kuprys [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 22, 2007 2:49 PM
Subject: Re:
Hello, I am using the last version on subversion, I already configured the
file sip.conf:
[6614]
username=6614
type=friend
secret=*
qualify=no
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
context=meuvoip
callerid=6614 [EMAIL PROTECTED]
[6617]
username=6617
type=friend
secret=*
But, as shipped, asterisk doesn't have native fax support,
but it can
be patched in via the spandsp code, giving you 2 new
applications: RxFax and
TxFax. You can plumb an incoming answered call to RxFax and it will
decode the incoming fax stream into a TIFF file which you can
Has anyone gotten this configured to show all extensions vertically instead of
filling up the window. If so would you mind sharing your configuration
Yes I have tried searching terms like +asternic +op_panel +vertical and a slew
of others. Unsucessful though.
--
I've reviewed the bugs reports. I didn't see anything that applied to
this. Have you? Could you point it out to me?
Olle E Johansson wrote:
21 feb 2007 kl. 15.50 skrev James Fromm:
Anybody seen this behavior?
To determine if it's my config or a bug, could I trouble someone
running
so when I go to start asterisk... this message is showed:
[EMAIL PROTECTED] ~]# asterisk -rd
asterisk -h
Asterisk 1.2.14, Copyright (C) 1999 - 2005, Digium, Inc. and others.
Usage: asterisk [OPTIONS]
Valid Options:
-V Display version number and exit
-C configfile Use an
You have to actually start asterisk with a command like safe-asterisk
before you can connect to the console with the command asterisk -r
Mhayk Whandson wrote:
Hello, I am using the last version on subversion, I already configured the
file sip.conf:
[6614]
username=6614
type=friend
Nevermind, I found it. I'll put up an SVN version in my dev environment
today.
Thanks.
James Fromm wrote:
I've reviewed the bugs reports. I didn't see anything that applied to
this. Have you? Could you point it out to me?
Olle E Johansson wrote:
21 feb 2007 kl. 15.50 skrev James
interesting!
so it means, that you can now see caller id names between sip phones
connected to asterisk and phones connected to pbx?
PJ
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I am using a Cisco-2,811 router with PRI as a gateway between Asterisk and
Nortel TX-1. I had problems
interesting!
so it means, that you can now see caller id names between sip phones
connected to asterisk and phones connected to pbx?
Yes!
__Yehavi:
___
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asterisk-users
ahh! I am having this same problem all of a sudden
I've installed many TDM cards before ..never had this problem
what gives?
Trying to load zaptel 1.0.10 ...
Rev. G card ... tried uncommenting the revH fix in zconfig.h ...but no go
ideas?!
--
Chris
Michael C. Cambria [EMAIL PROTECTED]
Hello,
I am running version 1.4 with realtime support. I've set (for Snom phones
300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the
database).
- When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem
was that when such a phone received a call
On Thu, Feb 22, 2007 at 09:01:47AM -0600, Joe Dennick wrote:
You have to actually start asterisk with a command like safe-asterisk
like asterisk . safe_asterisk only adds noise and complication and
doesn't help you with anything.
BTW: it may also not be recommended to use -p at start, is it
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look at
wiki...
exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten = s,2,Dial(SIP/myphone)
Or
Hi! Everyone,
Has anybody, already experienced any issue with port forward and NAT
translation into the CheckPoint Firewall?
I did setup a DMZ for asterisk be accessible from the LAN and WAN as well.
The needed ports were properly opened but even though I am not able to
authenticate into it from
Hi,
I'm currently working on a setup between Asterisk and VoiceGenie (which
is a IVR system).
The way my setup is done, is that I have a PRI line coming in my
Asterisk server, and then VoiceGenie is connected to Asterisk via SIP,
like any other softphone basically. I'm able to receive calls
On Thu, 22 Feb 2007 13:46:07 -0200, Alcides [EMAIL PROTECTED] wrote:
Hi! Everyone,
Has anybody, already experienced any issue with port forward and NAT
translation into the CheckPoint Firewall?
I did setup a DMZ for asterisk be accessible from the LAN and WAN as well.
The needed ports
I just wanted to share the solution for this problem. The busydetect
feature is working with all cell phone carriers now as well. I added
the following to my Zapata.conf. rxgain=4.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor G.
Hammonds
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look at
wiki...
exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten =
22 feb 2007 kl. 16.38 skrev Yehavi Bourvine +972-8-9489444:
Hello,
I am running version 1.4 with realtime support. I've set (for
Snom phones
300/320/360) a call limit of 1 (incominglimit and outgoinglimit
fields in the
database).
- When I used 1.4 SIP SHOW PEER show that it has a call
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
My point is that if it's going to involve rebuilding a kernel to support
IO-APIC, then I'd just as soon build from the ground up.
And my point is that this is the Asterisk Users mail list, not the
Trixbox list. Either ask other there or ask
Mindaugas Kuprys wrote:
Try to see maybe it could be done with Patton Smartnode series gateway.
Thank you. I take a look at the SmartNode 1200 [1]. It's great, but I don't
known too much about ISDN technologies and I'm not sure if this device will work
with a ISPBX line.
As far as I known, the
George Camilleri wrote:
Try http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways
Thanks. I visited this reference before, but I don't known if this gateways can
work with my current configuration.
I further describe my problem in other post.
--
Atentamente,
On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote:
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
I could be wrong here, but I think one of the symptoms of that could
be not have any zaptel devices and not having
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
http://astrecipes.net/index.php?n=248
I know it isn't everybody's piece of cake, but I thought
Not sure about * 1.4, but you can definitely use our Qloaderd script to do
that - see http://queuemetrics.com/download.jsp . That script is pretty
smart (to be a loader script...) and is able to handle restarts and
database disconnections.
l.
In data Thu, 22 Feb 2007 09:20:59 +0100,
Lacy Moore - Aspendora wrote:
On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote:
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
I could be wrong here, but I think one of the symptoms of that could
be not have
I am planning to develop an open source (GPL) queue statistic/analyzer.
Can i use that to store data into the db?
Or shall i wrote some php code to do that?
On 2/22/07, lenz [EMAIL PROTECTED] wrote:
Not sure about * 1.4, but you can definitely use our Qloaderd script to do
that - see
From: Olle E Johansson [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:36:45 +0100
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command? Thanks for
pointers.
None.
What do you want to do with SIP INFO?
/O
I was watching the send variable thread
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:09:07 -0600
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look
From: lenz [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 17:30:44 +0100
Hello list,
I have prepared a small tutorial today that deals with how to avoid
Asterisk rebuilding DTMF tones when using it to connect industial
appliances that use DTMF. You can find it at:
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can be solved using phone autoanswer feature, look at
wiki...
exten
I am programming a very large dialplan right now (Asterisk 1.4), and a
couple of things are annoying the heck out of me.
1. When in a macro, background() does not work properly. If you use the
background() app inside a macro, and then press a key, execution returns
back to the calling context
Chris Earle wrote:
agent monitoring screen?
curious,
which app are you using for that?
Unfortunately a proprietary app written in a closed language called
Progress.
However, the basics are that we embedded the Ipworks ActiveX xmpp
control, created a text box for each agent of a queue,
Yuan LIU wrote:
From: Olle E Johansson [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:36:45 +0100
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command? Thanks for
pointers.
None.
What do you want to do with SIP INFO?
/O
I was watching the
Well, kind of - it is meant for weird situations where mostly you do not
have regular POTS phones. Of course all DTMF detection would be disrupted.
l.
In data Thu, 22 Feb 2007 18:59:50 +0100, Yuan LIU [EMAIL PROTECTED] ha
scritto:
From: lenz [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007
From: Doug Garstang [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:17:20 -0800
I am programming a very large dialplan right now (Asterisk 1.4), and a
couple of things are annoying the heck out of me.
I have not programmed large dial plans, but have encountered some of the
nuances.
1. When in
Does anyone know why when calling out with an ATCOM AG-188 registered with
IAX (haven't tried SIP), there is no ring.
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asterisk-users mailing list
To UNSUBSCRIBE or update options
Ray,
I have been playing with OpenPBX. My core servers are Asterisk so I was
playing around with their T38Gateway application. Long story short - I can get
the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the
gateway feature of that product is still under
From: Philipp Kempgen [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 19:34:37 +0100
Yuan LIU wrote:
From: Olle E Johansson [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:36:45 +0100
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command? Thanks
for
On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote:
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 09:39:22 +0100
I think, this can
We have a * box with sip in, and h.323 out. When the H.323 call setup is
underway, will Asterisk translate the progress/status/result codes to SIP
automatically?
Ordo we have create our own result codes in SIP headers?
Thanks,
MD
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Any benefit on getting the PCI Express version?
Bill
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Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State toggled).
But it would be great to get an optical feedback of that phone's state
regarding the queue membership.
Does someone know if it is possible to
On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote:
Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State toggled).
But it would be great to get an optical feedback of that phone's state
On 2/22/07, Derek Whitten [EMAIL PROTECTED] wrote:
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
check the date on the machine?
Also check if you are running a NTP client, either ntpd or a periodic call
to
22 feb 2007 kl. 19.34 skrev Philipp Kempgen:
Yuan LIU wrote:
From: Olle E Johansson [EMAIL PROTECTED]
Date: Thu, 22 Feb 2007 10:36:45 +0100
21 feb 2007 kl. 21.58 skrev Yuan LIU:
What Asterisk command I can use to send a SIP INFO command?
Thanks for
pointers.
None.
What do you want to
On 2/22/07, Norbert Zawodsky [EMAIL PROTECTED] wrote:
Does someone know if it is possible to light up a LED under this szenario?
1.2 or 1.4?
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How does one configure asterisk for using TCP transport for SIP and not UDP?
Thanks, Jerry
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SIP has the following features:
·Lightweight, in that SIP has only six methods, reducing
complexity.
·Transport-independent, because SIP can be
22 feb 2007 kl. 21.02 skrev Bill Gibbs:
Ray,
I have been playing with OpenPBX. My core servers are Asterisk so
I was playing around with their T38Gateway application. Long story
short - I can get the ATA (behind NAT) to talk T38 to the rxfax app
on an OpenPBX server but the gateway
22 feb 2007 kl. 22.30 skrev Sune Kloppenborg Jeppesen:
On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote:
Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State
toggled).
But it would be great
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