What exactly do you mean by crossing channels? Mixed audio? Can
callers hear each other?
Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo:
Hi,
I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior,
because sometimes they are crossing channels, thus producing unwanted
calls
Le 18/02/2013 18:54, Chris Bagnall a écrit :
On 18/2/13 5:39 pm, Administrator TOOTAI wrote:
on incoming call we have exten =
100,n,Dial(SIP/Handset_102SIP/Handset_103SIP/Handset_104,,)
and always only Handset_102 is ringing, we receive busy back from the
2 others but they are not. Any clue?
All,
I'm trying to send an SMS directly from asterisk but it doesn't seem to be
working. The SMS() function does create an outgoing file but doesn't deliver
the SMS. Can anyone help me to understand how SMS() works. Thanks.
extensions.conf example:
same = n,SMS(hello,a,17654307001,hello
El 19/02/13 03:59, Thorsten Göllner escribió:
What exactly do you mean by crossing channels? Mixed audio? Can
callers hear each other?
Am 19.02.2013 02:07, schrieb Juan Carlos Agudelo:
Hi,
I have installed Asterisk 1.6.2.17-rc2 and I have a strange behavior,
because sometimes they are
On Tue, Feb 19, 2013 at 9:14 AM, Nicholas Johnson nejohns...@me.com wrote:
All,
I'm trying to send an SMS directly from asterisk but it doesn't seem to
be working. The SMS() function does create an outgoing file but doesn't
deliver the SMS. Can anyone help me to understand how SMS()
- Original Message -
From: David C Klaverstyn david.klavers...@intergraph.com
Is it possible to display the incoming calling number on a handset
when trying to pick up a call from another handset?
I currently have Call Pickup working using *8, I have also used the
PickUp
Check out connectedline()
-Original Message-
From: Rusty Newton rnew...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 19 Feb 2013 09:58:30
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing
Exactly, mixed audio, callers are linked to the call of another
caller,the calls are interlaced, is something that happens sometimes...
It can happen with analog dahdi calls. If this is the case, start
inbound on one end of the group, outbound from the other end.
Adrian
--
I don't have analog channels, this happens with SIP Trunk...
Juan..
El 19/02/13 11:06, Adrian Serafini escribió:
Exactly, mixed audio, callers are linked to the call of another
caller,the calls are interlaced, is something that happens sometimes...
It can happen with analog dahdi calls. If
Thanks for the help. Right now I'm running asterisk on a raspberry pi using a
phone number from flowroute. Is using a company like flowroute the same as
connecting to the PSTN? Also i've tried to install smsq but I couldn't find
any good documentation to get it setup properly. So no, I'm
On Tue, Feb 19, 2013 at 10:12 AM, Nicholas Johnson nejohns...@me.comwrote:
On Feb 19, 2013, at 10:41 AM, Christopher Harrington wrote:
On Tue, Feb 19, 2013 at 9:14 AM, Nicholas Johnson nejohns...@me.comwrote:
All,
I'm trying to send an SMS directly from asterisk but it doesn't seem to
be
On 02/19/2013 11:20 AM, Christopher Harrington wrote:
I was always under the impression you needed to either use a cellphone
type device to send them using your account, or send on to one of the
aggregators who have apis for this.
For low volume stuff, you can simply send an email
On Tuesday 19 February 2013, Nicholas Johnson wrote:
Thanks for the help. Right now I'm running asterisk on a raspberry pi
using a phone number from flowroute. Is using a company like flowroute
the same as connecting to the PSTN? Also i've tried to install smsq but I
couldn't find any good
2013-02-19 17:10, Juan Carlos Agudelo skrev:
I don't have analog channels, this happens with SIP Trunk...
Juan..
I've seen this with one of our sip-trunks. Our provider used opensips
for that platform I think. If had the same account registered in two
asterisk-servers and they answered the
-Original Message-
From: A J Stiles asterisk_l...@earthshod.co.uk
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Hello everyone, I’m new to Asterisk and I have a question. There is a phone
call between two users, then they are talking to each other directly or by the
server. I mean all packets from the user A to user B will be send directly to
each other or will those packets from user A must be send to
On Wed, 20 Feb 2013, Nguyễn Công wrote:
There is a phone call between two users, then they are talking to each
other directly or by the server. I mean all packets from the user A to
user B will be send directly to each other or will those packets from
user A must be send to server and server
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