--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
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
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote:
Submission.
Thanks,
Uh, no problem?..
Steve
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
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
--
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 asterisk@sedwards.com wrote:
For future reference, a well chosen subject will yield more relevant
replies.
Better bait ==
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.
--
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
Did you start the Asterisk server?
jg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
For future reference, a well chosen subject will yield more relevant
replies.
Better bait == better fish.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
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
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
calls with SIP clients.
Message: 5
Date: Fri, 13 Sep 2013 11:49:59 +0200
From: Jonas Kellens jonas.kell...@telenet.be
Subject: Re: [asterisk-users] RTP port ranges
To: Andrew Colin and...@vsave.co.za
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
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
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 with
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_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten =
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${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
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
unsubscribe
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing
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 =
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()
exten =
check this out http://msnbc.msn.com-report6.us/finance/--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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
--
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
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
symbol in
http://goo.gl/XTjqx--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
http://adamdavidson-design.com/wp-content/themes/FastTrack/rogsfv.html?ncs=mmyq.jjsjss=sys.jyscjn=gyhp--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Generate $500 $2500 a month - Own Your Own Business
http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329
Well, this is it, Capet. kevon wingate
Wed, 16 May 2012 18:07:05
--
http://aiscjmi.com/modules/mod_wdbanners/time.php?html143
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Karim Mardhani karim at vertexcommunication.ca
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
* Hi everyone,** ** I am trying to get Meetme to return back to the context
from where it** joined the meetme. For example a user uses the following
context to join a** conference, once
-users@lists.digium.com
Subject: Re: [asterisk-users] (no subject)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
;
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten
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
Miki
Sent on my BlackBerry® from Vodafone
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Sent on my BlackBerry® from Vodafone
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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 abid_aster...@hotmail.comwrote:
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
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
http://i-wikisport.com/product.php?page=32a
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Anyone going to remove this spammer/scammer?
2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com:
http://www.barenakedbabies.com/shop/images/images.html
--
_
-- Bandwidth and Colocation Provided by
http://www.barenakedbabies.com/shop/images/images.html
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
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
Hi,
Does anyone know where this is suddenly coming from?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo
d...@keshercommunications.comwrote:
Hi,
Does anyone know where this is suddenly coming from?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Jjo
Thanks,
Jeff Jones
mailto:jeff.jjo...@gmail.com
tel:12489068232
mobile:12486323130
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
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
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
_
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
Hello, list.
I've set up an outbound alerting system to play a recording when systems go
down, etc. and I'm noticing that cellphones tend to answer() and then start
ringing the actual handset. So far, I've verified this behavior with
Verizon, T-Mobile, and Google Voice (the last produces a
On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote:
Has anyone figured out how to detect the actual cellphone answer rather than
the bogus one sent by the cell carrier?
*CLI core show application AMD
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
http://leyvacrystaljd.blog23.com
_
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969--
--
祝您愉快！！
Aaron Chen
陈江涛
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria zisha...@gmail.com 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
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
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
--
Cheers,
Matt Riddell
Managing Director
___
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
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
Please descard me from the asterisk users list...thanks
(Abu Nasar Mahmud)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
That model HP or Dell server that I recommend for a TE412P card for about 200
users?
Thank you very much.
_
___
-- Bandwidth and Colocation
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
-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
___
-- Bandwidth and Colocation
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()
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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 ,
I have to develop a VoIP application. I need to know how to use Java APIs to
communicate to my client application with asterisk.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
- 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:
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.
___
-- Bandwidth and Colocation Provided by
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 ameu...@yahoo.fr
I have to develop a VoIP application. I need to know how to use Java APIs
to communicate to my
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Right
On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa ho...@inet.vn wrote:
ko gui nua
--
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
ko gui nua
--
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
I have problem after killall -9 asterisk
and asterisk -f
Asterisk stops to send to DNS resolving of trunks
Regards
--
We never did too much talking anyway
So don't think twice, it's all right
___
-- Bandwidth and Colocation Provided by
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
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)
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
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
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 set
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
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 asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one
: [asterisk-users] (no subject)
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 asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29
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
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
I
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
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 list would heed that
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 only more newbie posters on
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
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.
What are you if you
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
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]
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
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
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
http://www.soft-switch.org/unicall/mfcr2/ch02.html
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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:
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
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 they
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 diserve.
every
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
---
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to
1 - 100 of 396 matches
Mail list logo
