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Hi,
Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is
working fine but i cannot dial out. i don't hear anything on the phone and
asterisk CLI also does not show anything. my config is. please advice.
[2001]
type=endpoint
context=out-local
disallow=a
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota wrote:
> Submission.
>
> Thanks,
Uh, no problem?..
Steve
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Submission.
Thanks,
Francisco Leonardo Mota
Analista de Operações
DAGSer - Diretoria Adjunta de Gestão de Serviços
RNP – Rede Nacional de Ensino e Pesquisa
Site:http://www.rnp.br
Tel.:+55 61 3243-4384
Cel.:+55 61 9189-6660
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If you're using a redhat based distro, have you checked SELinux? Try
disabling (will require a server reboot)
Regards
Ish
On 3 September 2014 20:41, Steve Edwards wrote:
> For future reference, a well chosen subject will yield more relevant
> replies.
>
> Better bait == better fish.
>
> --
>
For future reference, a well chosen subject will yield more relevant
replies.
Better bait == better fish.
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-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
Did you start the Asterisk server?
jg
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To: asterisk-users@lists.digium.com
Subject: [asterisk-users] (no subject)
Hello asterisk-users,
Just compiled and installed 11.12.0 however when I try to connect with
rasterisk I get:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
It seems that asterisk.ctl
Hello asterisk-users,
Just compiled and installed 11.12.0 however when I try to connect with
rasterisk I get:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
It seems that asterisk.ctl is not created.
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Dahdi on Archlinux
I was able to compile the latest 2.9 Dahdi in archlinux on the Beaglebone black
without errors. I ran make install and make config. It installed the modules
etc correctly but did not create an init script in systemd or anywhere else.
Has anyone else been able to get dahdi to
Hi, all
I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded
from asterisk.org). We named it "Asterisk11".
I want to generate a call file to /var/spool/asterisk/outgoing. This call
will dial out to Local Channel and return to some Extens.
Then Asterisk11 will generate a CDR rec
To Jonas:
I have an asterisk box at home and I have this line in my rtp.conf file:
rtpstart=1
rtpend=10100
And My FW is setup to forward all incoming ports of range 1-10100 to
the asterisk PC.
I've never had a problem since one year, but I have never received more
than two simultaneous
Hi
I am running following asterisk installed with apt on Debian 7.1.
dpkg -l |grep asterisk
ii asterisk 1:1.8.13.1~dfsg-3+deb7u1
amd64Open Source Private Branch Exchange (PBX)
ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1
all Configura
thanks for your response
with the code below i can't get the extenssions 223
exten => 529,1,Answer()
exten =>
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten => 529,n,Dial(SIP/223)
exten => 529,n,Hangup()
i can get my number only wi
Define it as a variable, use the variable to define the filename
Ex.
exten =>
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten => 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to re
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten => 529,1,Answer()
exten => 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten => 529,n,Dial(SIP/223)
exten => 529,n
hello all,
i want to have ooh323 connection between asterisk and cisco. in my
scenario, asterisk is gateway and cisco is gatekeeper.
this is my ooh323.conf file:
[general]
port=1720
bindaddr=192.168.0.227
gateway=yes
faststart=yes
h245tunneling=yes
h323id=g...@test.com
settracelevel=10
gatekeeper=1
unsubscribe
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On Friday 12 April 2013, Thomas Perron wrote:
> Basic Dial Plan
>
> Why is this plan not engaging the line
> exten => 105,n,Dial(SIP/voipvoip.com/1703501)
> and dialing the 703 number?
>
> The logs and debug dont show any problems
>
>
> [incoming]
> exten => 44,1,Answer()
> exte
Basic Dial Plan
Why is this plan not engaging the line
exten => 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?
The logs and debug dont show any problems
[incoming]
exten => 44,1,Answer()
exten => 44,n,Wait(1)
exten => 44,n,Playback(beep)
exten =
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Thanks ajs
On Monday, July 30, 2012, A J Stiles wrote:
> On Monday 30 July 2012, akhilesh chand wrote:
> > Hi,
> > I'm not able to configure 8 port card whenever I configure it is showing
> > fatal: error inserting
> > wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
> > symb
On Monday 30 July 2012, akhilesh chand wrote:
> Hi,
> I'm not able to configure 8 port card whenever I configure it is showing
> fatal: error inserting
> wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
> symbol in module, or unknown parameter
It sounds as though you need to r
Hi,
I'm not able to configure 8 port card whenever I configure it is showing
fatal: error inserting
wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
symbol in module, or unknown parameter
Please help.
Regards
Akhilesh
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Generate $500 $2500 a month - Own Your Own Business
http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329
Well, this is it, Capet. kevon wingate
Wed, 16 May 2012 18:07:05
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Karim Mardhani http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote:
>* Hi everyone,*>* *>* I am trying to get Meetme to return back to the context
>from where it*>* joined the meetme. For example a user uses the following
>context to join a*>* conference, once user hangs up I would l
ent from BlackBerry® on Airtel
-Original Message-
From: Sam Govind
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 7 Sep 2011 11:53:33
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
;
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten => _X.,1,Wait(${INCOMING_WAIT})
exten =>_X.,2,Verbose(TB)
exten =>_X.
We are having several issues with call parking in Asterisk 1.8.5.
First, when a call is parked it is announcing the park location to the
caller rather than the callee. We also are experiencing an issue
whereby if you attempt to retrieve a parked call when a new call is
incoming the new caller and
Sent on my BlackBerry® from Vodafone
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as
Miki
Sent on my BlackBerry® from Vodafone
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Good morning gentlemen, is my first post in the list, now I'm starting asterisk
wanted to have your help for some questions.
Well the first function is as follow me. Here
I will demonstrate how this configuration follow me on my
extensions.conf but it is not working, and do not know why, but
you running GSM FWTs with asterisk ?
On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem wrote:
> HI,
>
> I am trying to setup a Class 4 termination setup using a kind of channel
> hunting scenerio. I have some SIP DID numbers assigned from the local
> telecom provider for termination. MY call comes fr
HI,
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local telecom
provider for termination. MY call comes from my wholesale client and lands on a
switch, then it is routed to asterisk. I want asterisk to ro
http://i-wikisport.com/product.php?page=32a
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Anyone going to remove this spammer/scammer?
2010/12/19 Dmitry Kupchinetsky :
> http://www.barenakedbabies.com/shop/images/images.html
>
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http://www.barenakedbabies.com/shop/images/images.html
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Hi,
I want to know that I have created a IAX2 trunk between two trunk I am
observing a packet rate of 50packet/sec mean packetization time=20ms but I want
to know that how to change the packetization time I have placed "trunk freq=50"
in general section of IAX but can not see any difference a
On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo
wrote:
> Hi,
>
>
>
> Does anyone know where this is suddenly coming from?
>
>
>
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Hi,
Does anyone know where this is suddenly coming from?
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Jjo
Thanks,
Jeff Jones
mailto:jeff.jjo...@gmail.com
tel:12489068232
mobile:12486323130
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Hi,
I have a problem with a SIP trunk between Asterisk and central OXE Alcatel,
especially sometimes are not received inbound calls with following messages:
-- Executing [...@test:1] AGI("SIP/800-084250f8",
"agi://127.0.0.1/test.agi") in new stack
-- AGI Script agi://127.0.0.1/te
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Friday, July 16, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] (no subject)
Ok I have a
Ok I have a queue that is working perfectly.
The problem is when one of the agents is outside the building on an
external phone line (say a cell phone). My telco hangs up on the call .
I think the telco is hanging up on these calls because there is no CID
attached. (I know my telco wont connec
http://leyvacrystaljd.blog23.com
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陈江涛
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On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria wrote:
> Fail2ban is a must. I was a victim of such attacks, and have implemented
> some other measures too, but fail2ban is a must have with the link posted by
> Matt which describes how to set it up for asterisk. Make sure you put your
> own ip ad
Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put your
own ip address in ignore list otherwise it can block you too.
On 2010-03-
On Fri, 19 Mar 2010, Adrian Marsh wrote:
I’m looking for some advice on securing Asterisk.
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate betwee
On 19/03/10 1:19 PM, Adrian Marsh wrote:
> Hello,
>
> I’m looking for some advice on securing Asterisk.
Have a look at fail2ban:
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
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Matt Riddell
Managing Director
__
Hello,
I'm looking for some advice on securing Asterisk.
Recently my servers been under several brute-force SIP attacks.
I have several remote sites, as well as many roaming users, who may have
PC softclients and/or SIP based hardphones.
My first step will be to strengthen the password
If you read your message all the way to the end, and every posting, you
will discover exactly how to do that on your own.
asterisk-users mailing list To UNSUBSCRIBE or update options visit:
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-
Please descard me from the asterisk users list...thanks
(Abu Nasar Mahmud)
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Hi,
That model HP or Dell server that I recommend for a TE412P card for about 200
users?
Thank you very much.
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On Tue, 20 Oct 2009, mickael ropars wrote:
> I want to know if it's possible to create a log file per context? and
> each time a context is restarted a ne x log file is created.
This is not clear to me. Contexts are not "restarted." What are you trying
to log?
Asterisk has the system() applica
terisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] (no subject)
All,
I want to know if it's possible to create a log file per context? and each
time a context is restarted a ne x log file is created.
regards
Mickael
_
All,
I want to know if it's possible to create a log file per context? and each
time a context is restarted a ne x log file is created.
regards
Mickael
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Hi
I use dial with music on hold command
exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , t
use ami
http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Java
or
Ajam
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM)
2009/3/19
> I have to develop a VoIP application. I need to know how to use Java APIs
> to communicate to my client applicati
On 19 Mar 2009, at 15:08, ameu...@yahoo.fr wrote:
> I have to develop a VoIP application. I need to know how to use Java
> APIs to communicate to my client application with asterisk.
Ok.
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- ameu...@yahoo.fr wrote:
>
> I have to develop a VoIP application. I need to know how to use Java APIs to
> communicate to my client application with asterisk.
I tried looking for some answers based upon your subject but nothing came up.
This may be what you're looking for: http://lmgtfy
I have to develop a VoIP application. I need to know how to use Java APIs to
communicate to my client application with asterisk.
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Right
On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa wrote:
>
>
> ko gui nua
> --
>
>
>
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Hello,
I have problem after killall -9 asterisk
and asterisk -f
Asterisk stops to send to DNS resolving of trunks
Regards
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What asterisk cli shows when you soft hangup these channels
Shariq
On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen <[EMAIL PROTECTED]>wrote:
> V 1.4
>
> When I do a "show channels" I get the following.
>
> CLI> show channels
> Channel Location State Application(Data)
>
V 1.4
When I do a "show channels" I get the following.
CLI> show channels
Channel Location State Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
Page(&
Hi All,
I have one doubt, suppose we have conference between 3
users (PCM
companded voice channels) then we add the streams together with scaling but
data which a user can receive will include his own voice information also
or i think we should substract his info. from the
Hi -
> I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm
> using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk.
> When I try to build Asterisk this is the error I'm getting.
>
> src/add.c:1: error: CPU you selected does not support x86-64 instruction se
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using
CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I
try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
I just
Alex Balashov wrote:
) How about rejecting emails that don't have a subject?
That is an excellent idea.
If a person doesn't have enough clue to use a subject, then we're really
just feeding the beast when we indulge the question with an answer.
And the archived version of that question/an
On Fri, 4 Jul 2008, Peter Lindquist wrote:
>> Steve Edwards wrote:
>>> But deciphering posts from our non-English-speaking members is half the
>>> challenge/fun :)
>>>
>>> Seriously though, good for them for trying. I wouldn't.
>>>
>>> What are you if you speak 3 languages? Trilingual.
>>>
>>
Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communicat
Steve Edwards wrote:
> On Thu, 3 Jul 2008, Alex Balashov wrote:
>
>> Steve Edwards wrote:
>>> On Thu, 3 Jul 2008, Alex Balashov wrote:
>>>
C F wrote:
> The number one skill for setting up asterisk is learn how to
> communicate since it's a communication application :P
Oh, if
On Thu, 3 Jul 2008, Alex Balashov wrote:
> Steve Edwards wrote:
>> On Thu, 3 Jul 2008, Alex Balashov wrote:
>>
>>> C F wrote:
>>>
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
>>> Oh, if only more newbie posters on this
Steve Edwards wrote:
> On Thu, 3 Jul 2008, Alex Balashov wrote:
>
>> C F wrote:
>>
>>> The number one skill for setting up asterisk is learn how to
>>> communicate since it's a communication application :P
>> Oh, if only more newbie posters on this list would heed that advice.
>
> ) How about rej
On Thu, 3 Jul 2008, Alex Balashov wrote:
> C F wrote:
>
>> The number one skill for setting up asterisk is learn how to
>> communicate since it's a communication application :P
>
> Oh, if only more newbie posters on this list would heed that advice.
) How about rejecting emails that don't have a
C F wrote:
> The number one skill for setting up asterisk is learn how to
> communicate since it's a communication application :P
Oh, if only more newbie posters on this list would heed that advice.
do u rely think this iz an acceptbl manner o/discoorse?
--
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Evariste Systems
Web
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
As for your problem looks like you are trying to use the wrong span
for dial out.
On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya <[EMAIL PROTECTED]> wrote:
>
>
> Hello everybody
>
>
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
loadzone = in
defaultzone
d inband dtmf 1 but says nothing about 9.
Am I doing anything wrong in the extension.conf.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob
Sent: Thursday, July 03, 2008 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia <[EMAIL PROTECTED]> wrote:
> From: Neha Punia <[EMAIL PROTECTED]>
> Subject: [asterisk-users] (no subject)
> To: "asterisk-users@lists.digium.com"
> Date: Thursday, July 3, 2008, 10:29 AM
> Hi
> I m maki
Hi
I m making a call from one asterisk server to an asterisk client
The call gets completed but I want it to send dtmf signals
The dialplan I have made for this is like
exten => 205,1,Answer
exten => 205,n,Wait(15)
exten => 205,n,Playback(dtmf-1)
exten => 205,n,Wait(20)
but it does not send any
Hi :
asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main
configured files are:
extensions.conf:
[from-pstn]
exten => 9711315,1,Dial(SIP/3000,30)
exten => 9711315,2,VoiceMail([EMAIL PROTECTED])
exten => 9711315,3,PlayBack(vm-goodbye)
exten => 9711315,4,HangUp()
sip.conf:
[3000]
the subject of this thread has been on this list way too many times
just search the archives.
On 5/23/08, Joseph L. Casale <[EMAIL PROTECTED]> wrote:
> In the setup tutorial @
> http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
> it states the potential issue regarding se
In the setup tutorial @
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
it states the potential issue regarding setting up UniqueID
as the primary key, but doesn't state how to rectify this?
What is the proper way to make sure this is done right?
Also, has anyone buil
I heard something about the agents.conf file in the asterisk pbx.. I would
love to have a tutorial or someone that will help me doing this.. it's not
working out with her
Can anyone help ? it's getting frustrating with teaching the agents to
logoff the queue everytime.. or even teaching the superv
On Mon, Apr 28, 2008 at 9:32 AM, Arthur <[EMAIL PROTECTED]> wrote:
>
> > Make sure you get a "helpful tech" on the phone. Many times they will
> > just dismiss you with "we cannot do that" even though they may be able
> > to.
>
> i always say if you pay your bills you should get the support you di
>
> Make sure you get a "helpful tech" on the phone. Many times they will
> just dismiss you with "we cannot do that" even though they may be able
> to.
i always say if you pay your bills you should get the support you diserve. &
every provider is almost always willing to help out his clients if
Again, a reply to my reply. Note to self: stop hitting send before
completing thoughts.
Maybe if you ask the telco to turn off the SLA blocking. It may not
solve the underlying issue but it may allow you to continue inbound
and outbound without service interruption providing it does not drop
an
This may be more helpful as far as Asterisk implementation. Sorry I
cannot be of more help, I have never dealt with this tech.
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
Thanks,
Steve Totaro
On Mon, Apr 28, 2008 at 9:06 AM, Arthur <[EMAIL PROTECTED]> wrote:
> http://www.soft-switch.org/
http://www.soft-switch.org/unicall/mfcr2/ch02.html
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Dear Steve,
We have installed Asterisk with Digium card TE110P , install MFC R2 connect to
PSTN (indonesia) using DIG13 MFCR2 siemens EWSD, Germany.
asterisk working normaly, outgoing call ok, incoming call ok. but in central
office /PSTN having SLA(service level alarm). If It happend, all chann
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN->PSTN calls get this error...
-Greg
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