Hello Thelma,
Friday, February 16, 2018, 2:16:02 AM, you wrote:
> Contact: "sip:pstn-"
> And it found in sip.conf only:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
> Is perhaps the name effected by the special character "-" (dash) that is
> why it only matches "pstn" and
Le 16/02/2018 à 05:30, the...@sys-concept.com a écrit :
On 02/15/2018 04:49 PM, Joshua Colp wrote:
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
Thanks again for the hint.
Here is the output from asterisk.
The call is coming on Audocodes gateway from: pstn-
But
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer
Thelma
On 02/15/2018 07:16 PM, the...@sys-concept.com wrote:
>
> On 02/15/2018 04:49 PM, Joshua Colp wrote:
>> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>>
>>
>>
>>>
>>> Thanks again for the hint.
>>> Here is the output from asterisk.
>>>
>>> The call is coming on
On 02/15/2018 04:49 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
>
>
>>
>> Thanks again for the hint.
>> Here is the output from asterisk.
>>
>> The call is coming on Audocodes gateway from: pstn-
>>
>> But asterisk display:
>> Found peer
On Thu, Feb 15, 2018, at 7:46 PM, the...@sys-concept.com wrote:
>
> Thanks again for the hint.
> Here is the output from asterisk.
>
> The call is coming on Audocodes gateway from: pstn-
>
> But asterisk display:
> Found peer 'pstn-9998' for '7804715665' from 10.10.0.8:5060
>
> Why not
On 02/15/2018 04:08 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
>> On 02/15/2018 03:44 PM, Joshua Colp wrote:
>>> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
I'm using Audio-codes MP-114 unit and it has two public lines PSTN
On Thu, Feb 15, 2018, at 7:03 PM, the...@sys-concept.com wrote:
> On 02/15/2018 03:44 PM, Joshua Colp wrote:
> > On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
> >>
> >> IN audocodes setting I have:
> >>
On 02/15/2018 03:44 PM, Joshua Colp wrote:
> On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>>
>> IN audocodes setting I have:
>> "EndPoint Phone Number"
>>
>> Channel: 3phone number: pstn-
>>
On Thu, Feb 15, 2018, at 6:43 PM, the...@sys-concept.com wrote:
> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
>
> IN audocodes setting I have:
> "EndPoint Phone Number"
>
> Channel: 3phone number: pstn-
> Channel: 4phone number: pstn-9998
>
> When I am
I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports
IN audocodes setting I have:
"EndPoint Phone Number"
Channel: 3phone number: pstn-
Channel: 4phone number: pstn-9998
When I am calling " pstn-" the port number "Channel:3" lights up but
asterisk is showing
On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
>
> My sip provider gave me 2 numbers for the incoming call via pstn.
>
> nro1 = 12341234
> nro2 = 45674567
>
> I have a dialplan for each.
> if i put this on my dialplan:
>
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
>
> Works!
>
> But if i
It seems like your SIP provider is not sending and DID information, or that
the information is not being sent in the same format you are using in your
dialplan.
You can check this by looking at the SIP debug information for the inbound
calls and/or by checking with your SIP provider (that they
Hi,
My sip provider gave me 2 numbers for the incoming call via pstn.
nro1 = 12341234
nro2 = 45674567
I have a dialplan for each.
if i put this on my dialplan:
exten => s,1,Dial(SIP/1001)
exten => Hangup()
Works!
But if i put one of them:
exten => 12341234,1,Dial(SIP/1001)
exten =>
Longtime lurker, first time poster. :)
A client of mine is in need of having Asterisk record every call that comes
in from a specific incoming route. I've added the following lines to the
sip_additional.conf file, but no recordings are showing up in the
/var/spool/asterisk/monitor/ folder.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Hall
Sent: Friday, June 10, 2011 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Recording
Longtime lurker, first
Hi all,
Recently I have instaled one Digium TDM410 on my Asterisk. After instaled ,
I can do outgoing calls but I cant receive calls. I receive the following
messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654]
chan_dahdi.c: Got event 18 (Ring Begin)...[Sep
On 09/15/2010 07:20 AM, Flavio Miranda wrote:
Recently I have instaled one Digium TDM410 on my Asterisk. After
instaled , I can do outgoing calls but I cant receive calls. I receive
the following messages:
chan_dahdi.c: Got event 2 (Ring/Answered)...
[Sep 14 11:24:44] NOTICE[2654]
As Kevin said, you need to define an 's' extension where the calls will be
answered. Seems like you are using default configuration. Open file
'extensions.conf' in /etc/asterisk folder and look for context named
[default]. If it is not there, create one and add something under it, e.g.,
[default]
Ok. Problem solved .
Thank you very much!!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Wed, 15 Sep 2010 09:56:36 -0400
From: zisha...@gmail.com
To: kpflem...@digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] incoming
: [asterisk-users] Incoming call doesn't finish when internal
phone hangs up
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktikxafnxhbsws0ov4u5ht3yjbeevuh26vehrg...@mail.gmail.com
Content-Type: text/plain; charset=ISO
Hello guys,
I have this problem when a call is received in my PBX:
(Caller) -- (Redirecting Service) -- (E1 PRI) -- (Asterisk PBX) --
(Internal Phone)
Reception works fine, but when conversation finishes and the agent at
internal phone hangs up, the call at caller's side is still alive for
many
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
OFFICE1=DAHDI/13,,rtT
OFFICE2=DAHDI/14,,rtT
If I add this line:
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2})
only OFFICE1 rings.
If I
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
typo ...
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}${OFFICE2},,rtT)
Gareth Blades wrote:
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi
Is there any documentation on the CallWaitingRing?
Thanks
Dan
-Original Message-
From: Danny Nicholas da...@debsinc.com
Sent: 12 November 2009 14:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Incoming Call
Hello,
I have Asterisk set up with 6 extensions. When a call comes in, I use a
Dial command to call all the extensions together until someone picks up.
The problem is, when there is an incoming call and an extension is in
use, if the extension puts down the phone while the incoming call is
- Original Message -
From: Dan Journo
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 12, 2009 1:24 PM
Subject: [asterisk-users] Incoming Call Ring
Hello,
I have Asterisk set up with 6 extensions. When a call comes in, I use
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Incoming Call Ring
Hello,
I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial
command to call all the extensions together until someone picks up.
The problem is, when there is an incoming
For determining security risks, its specific to how your dialplan is set up.
If a person connects to your asterisk, what can they do? what happens? did
you set the incoming context to one with outgoing dialing rules?
Also for filtering calls, you'll probably want to either look at the
incoming sip
Hi All,
I'm having a bit of trouble with my new *NOW setup.
I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from
SimpleSignal.com. Outbound calling works great, but I'm having some trouble
with inbound calls.
First, we would get the the number you have dialed is not in
Hi, i managed to connect to Ericsson MD110 with PRI at last.
And made a successful call thru asterisk to ericsson.
But when i try to call from ericsson to asterisk i got an error on
asterisk side.
And i couldnt figure out why.
Here's my extensions.conf about incoming calls.
[DID_span_1]
include
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i managed to connect to Ericsson MD110 with PRI at last.
And made a successful call thru asterisk to ericsson.
But when i try to call from ericsson to asterisk i got an error on
asterisk side.
And i couldnt figure out why.
Here's my
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i managed to connect to Ericsson MD110 with PRI at last.
And made a successful call thru asterisk to ericsson.
But when i try to call from ericsson to asterisk i got an error on
asterisk side.
And i couldnt figure out why.
Here's my
- Oguzhan Kayhan oguzh...@bilkent.edu.tr wrote:
Hi, as far as i know 's' is wildcard for all calls because as i see on
asteriskgui it is written as 's' (CatchAll) which means redirect all
calls to that extension.
That is not correct. The 's' extension only matches analog calls (because
sorry i cant help you :-(
i only can sugest add another peer in sip.conf in one use only audio and in
the other one use only T38.
you should post it again whit a subject like T38 problem or please
help!! t38 problem.
David
Sorry again.
2009/2/27 michel freiha mich...@gmail.com
Dear
paste your sip.conf.
David
2009/2/26 michel freiha mich...@gmail.com
Dear All,
I have created an inbound context in SIP .conf that forward incoming call
to opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS
Dear David,
Please find on http://pastebin.com/m69b8559d my sip.conf file
Thanks a lot
On Fri, Feb 27, 2009 at 1:05 PM, David fire ddf...@gmail.com wrote:
paste your sip.conf.
David
2009/2/26 michel freiha mich...@gmail.com
Dear All,
I have created an inbound context in SIP .conf that
Dear All,
I have created an inbound context in SIP .conf that forward incoming call to
opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
voice codec to INVITE packet...It just contains T.38
Dera All,
I have the following scenario,
A customer dial a DID number...The call is routed to a PSTN GW that send the
call to asterisk...
On asterisk I created an AGI Script that send the call to an extension
registered on OpenSIPS server...
The extension is ringing successfully, but as soon as I
if is a codec problem start putting all the systems in the same codec and
disallow all other.
put for example all in alaw and disalaw all other includeing ulaw.
you can make calls from asterisk to the sip extencion registered in opensip?
check that you can start the call from a soft phone or you
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution
SNIP
I'm using sipgate.co.uk for incoming calls, but when I make a test
call from the PSTN, the call just dies without connecting to my
Astlinux box. (I'm monitoring asterisk console via 'asterisk
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution, and have had a read of 'Asterisk - The
Future of Telephony'. So you could say that I have at least tried to
RTFM as it were!
I've configured a couple of Asterisk instances on both Debian and
CentOS based
Hello,
From the netstat output my initial *guess* is that asterisk is listening
(udp/5060, udp/2727, among others). One way to tell for sure would be
to run 'lsof -i' which would show you the process associated with the
port.
As far as the call not reaching asterisk or being a firewall
On Sun, 13 Jul 2008, Chris Rowson wrote:
Hi, this is my first post to the list, but I have tried to search
elsewhere for a solution, and have had a read of 'Asterisk - The
Future of Telephony'. So you could say that I have at least tried to
RTFM as it were!
I've configured a couple of
On Tuesday 04 March 2008 06:48:43 am marek cervenka wrote:
hi,
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
i'm using flash operator panel now
but i want something without flash (maybe something in AJAX?)
thanks
On Tue, Mar 4, 2008 at 7:18 PM, marek cervenka [EMAIL PROTECTED] wrote:
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
ADM http://adm.hamnett.org/ can invoke browsers when a call arrives.
raj
___
-- Bandwidth
ASTassistant can do this as well. www.astassistant.com
-Scott
- Original Message -
From: Rajkumar S [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 05, 2008 5:48 AM
Subject: Re: [asterisk-users
hi,
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
i'm using flash operator panel now
but i want something without flash (maybe something in AJAX?)
thanks
---
Marek Cervenka
===
Hi,
I have asterisk register two users (client-1, client-2) with a SIP proxy.
I have the same two SIP client registered with asterisk. Now my dial plan
setup is such that any call from client-1/client-2 is forwarded to the SIP
proxy and the SIP proxy then takes the routing decision. Calls
- Original Message
From: Mayur [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, 31 January, 2008 9:59:42 AM
Subject: [asterisk-users] Incoming call from SIP proxy to asterisk
Hi,
I have asterisk register two users (client-1, client-2) with a SIP
Hi, I am a newbie putting together my first Asterisk system
and having a problem with the IVR handling incoming calls.
I installed the Asterisk Trixbox version 1.2.2 with a X100P
FXO PCI card. I have a PSTN line connected to the card. I set up
two extensions: 200 and 201. I created a
I have a problem :
when i receive a
call in h323 and send on zap channell, there is no cdr..
if i receive in sip
is all ok .
Why
??
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Hi,I have registered with IPKall ang got the number i.e., 206XXX. When I call to this number, It is telling that "The party you are calling is currently busy". Here I am giving my config details.When I registered with IPKall, I entered these below values:SIP Phone number: 7312567SIP Proxy:
Crazy Boy wrote:
I have given my total configuration. Please tell me the solution.
Looking forward to your response. Thank you.
You need to also include the output from the console.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Von: Crazy Boy [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 7. September 2006 14:25
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Incoming call problem-calling part is busy(IPKall)
Hi,
I have registered with IPKall ang got the number i.e., 206XXX. When I
call
I have this in sip.conf:
[ata1]
username = ata1
accountcode = ata1
qualify = yes
secret = foo
type = friend
host = dynamic
fromdomain = ipt.gumby.com
context = fromata_start
qualify = yes
When a call comes in from this device, if I have type=peer, Asterisk doesn't
match it, but it does if
You have a little confusion:
friend = can GENERATE and RECEIVE calls
peer = can only GENERATE calls
user = can only RECEIVE callsAlyed
Return-Path: [EMAIL PROTECTED] Fri Jul 07 09:27:13 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
My design is the following:
SS7_smgd[asterisk]Zap-backtoback---EuroISDN[Mediatrix box]Sip
customers
In this design I have a problem about early audio
no passed from asterisk to Zap un the incominga cal from Zap.
I am sure SS7_smgd passes "early audio" to Asterisk
because if I
: Friday, June 09, 2006 9:32 AM
Subject: [Asterisk-Users] incoming call
from Zap: "early audio" problem
My design is the following:
SS7_smgd[asterisk]Zap-backtoback---EuroISDN[Mediatrix
box]Sip customers
In this design I have a problem about e
Asterisk SVN-trunk-r7353M
I have a EuroISDN line. I am sometimes out of the office so I get my
extension to ring both my mobile and desk top (7960) phone at the same time.
This all works just peachy. However, I have a question regarding
callerid. Is there any way of setting the callerid so
Julian Lyndon-Smith wrote:
Asterisk SVN-trunk-r7353M
I have a EuroISDN line. I am sometimes out of the office so I get my
extension to ring both my mobile and desk top (7960) phone at the same
time.
This all works just peachy. However, I have a question regarding
callerid. Is there any way
Hi,
I am new to this group.I searched for my problem in the forum but could not find any solution. So here it goes:
In my work place we have an asterisk box. Everything works fine except
the fact that when I first call the work phone number from my cell the
auto-attendend works fine but If I
Hi,
I have already determine the reason why my incoming
got release after one ring. The telco that I am
connected is waiting for an immediate answer
supervision from my side. Is there anyway immediate
answer supervision be included on the ISDN messages.
Thanks
--- leonimar cape [EMAIL
Hello,
Can somebody please assist me with my problem.
Currently I am using a [EMAIL PROTECTED] version 2.4 with
a TE406P digium card. One the E1 is connected to a
telco switch via an ISDN. May issue is that may
incoming calls in the zap channels gets disconnected
or release after 1 ring. I really
[EMAIL PROTECTED]
i have a problem when hangup an incoming call,
i receive this error:
Incoming call: Got SIP response 503 Server error back from xxx.xxx.xxx.xxx.
and the caller stay connected and don't receive hangup
any idea?
___
--Bandwidth and
Which * version are you using ?
Isamar
On Tue, 17 Jan 2006, news.dalaidily news.dalaidily wrote:
[EMAIL PROTECTED]
i have a problem when hangup an incoming call,
i receive this error:
Incoming call: Got SIP response 503 Server error back from xxx.xxx.xxx.xxx.
and the caller stay
PROTECTED] *On Behalf Of *Health
Masters
*Sent:* November 15, 2005 7:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [Asterisk-Users] Incoming call trunk fwd not work
I just had that problem... I had to enter an extension with my fwd #
as the extension
,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Techsupport
Sent: November 21, 2005 5:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming call trunk fwd not work
in extensions.conf
[fromiaxfwd
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Health Masters
Sent: November 15, 2005 7:28 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Incoming call trunk fwd not work
I just had that problem... I had to enter an extension
with my fwd
[EMAIL PROTECTED]
www.k2systems.ca
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of AR Tarzi
Sent: November 15, 2005 8:35 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Incoming call trunk fwd not work
In AAH create
and such.. I
assumed that you had done that.
- Original Message -
From:
Cristian Paun
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Thursday, November 17, 2005
03:07
Subject: RE: [Asterisk-Users] Incoming
call trunk fwd not work
I
I have an AAH installed with trunk FWD. I am able to
place calls but not receive. I get these message Nov 15 13:05:52 NOTICE[1410]: Rejected connect attempt from
192.246.69.187, request '[EMAIL PROTECTED]' does not exist
My AAH box is in NAT mode
Can somebody give me a config file worked
I just had that problem... I had to enter an extension with my fwd # as
the extension under the context after @ and tell it what local
extension to ring in extensions.conf
;free world dialup incomming call
exten = 720727,1,Dial(IAX2/shawn, 30)
Cristian Paun wrote:
I have an
-Users] Incoming call
trunk fwd not work
I have an AAH installed with trunk
FWD. I am able to place calls but not receive. I get these message
Nov 15 13:05:52 NOTICE[1410]: Rejected connect
attempt from 192.246.69.187, request '[EMAIL PROTECTED]' does not
exist
My AAH box
Hello,
Just setup an asterisk server and Amp help me install their
portal and Asterisk. The server is up and all the hardware is
loading. Problem is with incoming calls. All calls go to the first
extension on the system and it still does not ring it goes straight to
voicemail. The
bump from last week
Hi all,
I have 12 SIP phones at a particular site all connected to a local asterisk
server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming
calls. An IAX gateway is used for outbound calls. At the moment, when an
incoming call comes in, asterisk dials every
Hi all,
I have 12 SIP phones at a particular site all connected to a local asterisk
server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming
calls. An IAX gateway is used for outbound calls. At the moment, when an
incoming call comes in, asterisk dials every SIP phone like so:
When I receive a call, only one telephone ring...
Can I receive a call in much telephones, therefore
more telephones rings?
___
Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB
http://mail.yahoo.it
Yes.
In cmd Dial you can specify multiple users/recieveres.
Ex.
Dial(SIP/usr1SIP/usr2)
The clue is .
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Fabio Montemaggiore
Sendt: 6. oktober 2005 10:06
Til: asterisk
Emne: [Asterisk-Users] Incoming
Hi!
Yes, you can call sa many phone you wantfor example:
100,1,Dial(SIP/firstPhoneSIP/secondPhone)
makes firstPhone and secondPhone SIP phones ring at the same time when
dialing extension 100.
Giorgio
Fabio Montemaggiore wrote:
When I receive a call, only one telephone ring...
Can I
6:57 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Incoming call #2 sent to VM immediately
whenalready on phone with incoming.
I'm having this problem where if the phone is ringing from IncomingCall
#1, IC#2 will be immediately sent to VM. Is there somethign wrong with
my dial
On Tuesday 09 August 2005 18:56, Min Hwan Chang wrote:
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
That's a good question. I have no idea. I'm pretty new at this, so I'm
just combining bits and pieces of what I find together. If anyone
could help, it'd be greatly appreciated.
On 8/10/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 09 August 2005 18:56, Min Hwan Chang wrote:
I'm
On Wednesday 10 August 2005 12:48, Min Hwan Chang wrote:
That's a good question. I have no idea. I'm pretty new at this, so I'm
just combining bits and pieces of what I find together. If anyone
could help, it'd be greatly appreciated.
Have you read through the Asterisk Handbook draft?
Hi all,
i've asterisk with 8 FXS module connected to 8 PSTN lines. Each line have it's
own number anche i want to do different action based on incoming call.
For example, if call is on Line 1 i want to redirect it to extension 203, on
line 2 to extension 201 etc etc
it's possible ? How ?
I'm having this problem where if the phone is ringing from
IncomingCall #1, IC#2 will be immediately sent to VM. Is there
somethign wrong with my dial plan? I currently have 4 incoming lines
going into a TDM400 with the group set to g0.
Could it be that the way I've set this up, if any of the
I am having a problem with your my nufone service.
I'm trying to setup incoming calls and I'm having no
success. Outgoing works fine though. The message I'm
getting is the person you are call is not currently
reachable. I'm going to give you as much info as I
can. I'm also an asterisk newb!
I am having a problem with your my nufone service.
I'm trying to setup incoming calls and I'm having no
success. Outgoing works fine though. The message I'm
getting is the person you are call is not currently
reachable. I'm going to give you as much info as I
can. I'm also an asterisk
On 6/8/05, Rick Baranowski [EMAIL PROTECTED] wrote:
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work all the time, the outgoing does not seem to have a
problem.
I am having the exact same issue... unfortunately they seem to be the
only people on
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using for the last several months:
[teliax]; for incoming calls
context=teliax-incoming
type=user
auth=md5
, June 09, 2005 5:59 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does
-Commercial
Discussion
Subject: Re: [Asterisk-Users] Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work
all the time, the outgoing does not seem to have a
problem.
Here's what I've been using
Title: Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the incoming call to work all the time, the outgoing does not seem to have a problem.
I have worked with their support but since they say that we are getting the initial call
Hi,
I have a sipura 2000 ATA connected to an asterisk server on the local
network, and the POTS line connected to asterisk using a X100P clone, when
calling remotely through the X100P (incoming call), the phone attached to
the sipura device always rings like it should, however sometimes it does
Hello All,
I would be grateful for some help with this issue. I have
gotten Asterisk to work with Broadvoice on outgoing calls only. When I try to
call my Broadvoice number from another location I get a caller is busy
message. My Asterisk box is behind a Linksys WRT54G. Could this be
@lists.digium.com
Sent: Monday, February 07, 2005 8:36
AM
Subject: [Asterisk-Users] Incoming Call
Problem
Hello
All,
I would be grateful for some help
with this issue. I have gotten Asterisk to work with Broadvoice on
outgoing calls only. When I try to call my Broadvoice number
Hi,
I have an broadvoice account that I am able to make, and recieve calls
on. There is just one minor issue. For incoming calls, the phone I
call from (not voip) does not ring while waiting for asterisk
extension to answer. This is not the case when I connect my SPA2000
directly to the
asterisk-users@lists.digium.com
Sent: Thursday, February 03, 2005 2:22 PM
Subject: [Asterisk-Users] Incoming call not ringing
Hi,
I have an broadvoice account that I am able to make, and recieve calls
on. There is just one minor issue. For incoming calls, the phone I
call from (not voip) does
Hello All,
Can asterisk play voice prompt and collect digits on the IP leg ( ie.
The incoming VoIP call)?.
Thanks Regards
-kts
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