2009/10/9 Juan E. Rodríguez jerdg...@gmail.com:
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work with Asteris.
/r
randulo schrieb:
2009/10/9 Juan E. Rodríguez jerdg...@gmail.com:
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
The Gigaset S675IP series of DECT/SIP phone has SMS capability, but
not sure it can work
Hi,
Using AMI, when a peer is set with Qualify=yes, it seems you can't make a
difference between First-time registration and Re-registration. Looking at
an AMI log, I saw:
Re-registration (to be confirmed):
Event: PeerStatus
Privilege: system,all
ChannelType: SIP
Peer: SIP/7266
PeerStatus:
2009/10/8 Leif Madsen leif.mad...@asteriskdocs.org
Please follow up on the issue tracker at http://issues.asterisk.org
Thanks!
Leif.
I think I will.
Two days ago we reverted back to 1.6.1.0 and Asterisk is running OK since.
Hopefully, if this behaviour remains, we will give 1.6.1.7-rc2 a
What I have tried is :
register = user1:pass...@server/yocan
register = user2:pass...@server/itcenter
extensions.conf :
[default]
exten = yocan,1,GoTo(user1,s,1)
exten = itcenter,1,GoTo(user2,s,1)
[user1]
...
[user2]
...
But the CLI shows :
[Oct 9 09:28:52] -- Executing
On Fri, Oct 9, 2009 at 2:18 AM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Thu, 2009-10-08 at 16:07 -0400, Michelle Dupuis wrote:
More specificallyI'm looking for a Linux package to allow shaping,
QoS, prioritization by port, etc.
snip
Spinning off from another
Hi,
One of the key features of Asterisk is that we can install it on many
hardware platforms.
We've done our best to script this installation process, so that, in case of
hardware failure, we can re-install Asterisk on another platform.
The question I have is how can we adapt our process so that
O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium
TE205P connected to a single span if ISDN PRI. The Telco has assigned
us two local numbers to test incoming calls. I created an inbound route
for one of those DID's and assigned it to one of our extensions. Sounds
simple
- Dovid Bender asteriskus...@dovid.net wrote:
| - Original Message -
| From: Trevor Peirce tpei...@digitalcon.ca
| To: Asterisk Users Mailing List - Non-Commercial Discussion
| asterisk-users@lists.digium.com
| Sent: Tuesday, October 06, 2009 23:14
| Subject: Re: [asterisk-users]
On Thu, Oct 08, 2009 at 10:00:19PM -1000, Ben Schorr wrote:
O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium
TE205P connected to a single span if ISDN PRI. The Telco has assigned
us two local numbers to test incoming calls. I created an inbound route
for one of those DID's
Hi all,
I have 2 question.
I have a call center queue with 5 agent; the following are the configuration
files:
*queue.conf*
[name_of_queue]
musicclass = default
announce = queue-name_of_queue
strategy = ringall
servicelevel = 60
context = callcenter
timeout = 60
retry = 5
wrapuptime=15
On Fri, 9 Oct 2009, Olivier wrote:
The question I have is how can we adapt our process so that Digium's G729
licences (or other licenced software) could be installed without asking too
long interactive sessions.
Download and deploy the free one.
Buy digium licenses to cover each anticipated
While we're on the subject of G.729...
I can end to end use it with no transcoding, but voicemail is the main
sticking point for me - I'd need to transcode.
So why can't voicemail store the audio in the format it's being streamed
in on?
Is there a technical reason for no voicemail storage in
Hello
Please let me know can we call normal PSTN lines as trunk lines?? As a
normal pstn line used in home .
One More thing that If i need ten PSTN lines on one Server then which Digium
card is suitable.
I am confused with TDM800P as it say it accepts a trunk line?
--
Best Regards
Shakeel
On Fri, Oct 9, 2009 at 10:37 AM, jonas kellens jonas.kell...@telenet.be wrote:
So the call comes into the right context... that's not the problem.
But my CDR is messed up. The accountcode that I have set for user1 is always
replaced for the accountcode I've set for user 2.
[YOCAN-3starsnet]
- Original Message -
From: --[ UxBoD ]-- ux...@splatnix.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 09, 2009 10:32
Subject: Re: [asterisk-users] MPG123 Dying
- Dovid Bender asteriskus...@dovid.net wrote:
I don't think there is much you can do since Asterisk matched it based on the
IP of your carrier. Maybe there is some sort of variable that you can set in
the dial plan ?
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent:
- --[ UxBoD ]-- ux...@splatnix.net wrote:
| - Dovid Bender asteriskus...@dovid.net wrote:
|
| | - Original Message -
| | From: Trevor Peirce tpei...@digitalcon.ca
| | To: Asterisk Users Mailing List - Non-Commercial Discussion
| | asterisk-users@lists.digium.com
| | Sent:
- Dovid Bender asteriskus...@dovid.net wrote:
| - Original Message -
| From: --[ UxBoD ]-- ux...@splatnix.net
| To: Asterisk Users Mailing List - Non-Commercial Discussion
| asterisk-users@lists.digium.com
| Sent: Friday, October 09, 2009 10:32
| Subject: Re: [asterisk-users] MPG123
We use 3Com managed gigabit switches that support QoS and priority for
VoIP.
3Com Unified Gigabit Wireless PoE Switch 24
and
3Com Baseline Switch 2924-PWR Plus
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Kenner
Sent: Thursday, October 08, 2009 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MeetMe option question
We've started to use
--- On Thu, 10/8/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
From: Tzafrir Cohen tzafrir.co...@xorcom.com
Subject: Re: [asterisk-users] No sound on voicemail from analog line
To: asterisk-users@lists.digium.com
Date: Thursday, October 8, 2009, 4:11 PM
On Thu, Oct 08, 2009 at
Dear all
i have a TE205P connected to an Asterisk 1.2.18.
Yes i know, the version is old but since now the system was stable and
i don't have the necessity of an upgrade.
The system provide an IVR service that:
1) receive the call
2) verify the queue length
3) hangup if queue length is 1
4)
Before we call each other liars and thieves, here is a link:
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
As with any open source, do your own homework on licensing, AND apply your
own reasonable judgment.
At this point I would like to confess that I watch rented DVD's under
- Dovid Bender asteriskus...@dovid.net wrote:
| - Original Message -
| From: --[ UxBoD ]-- ux...@splatnix.net
| To: Asterisk Users Mailing List - Non-Commercial Discussion
| asterisk-users@lists.digium.com
| Sent: Friday, October 09, 2009 10:32
| Subject: Re: [asterisk-users] MPG123
Gordon Henderson wrote:
While we're on the subject of G.729...
I can end to end use it with no transcoding, but voicemail is the main
sticking point for me - I'd need to transcode.
So why can't voicemail store the audio in the format it's being streamed
in on?
Why do you think it
Robert McGilvray wrote:
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller,
What's confusing me is that when I look in app_meetme.c, the relevant
options are stored in what are called conference flags and there are
separate user flags. This makes it
Richard Kenner ken...@gnat.com wrote:
Robert McGilvray wrote:
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller,
What's confusing me is that when I look in app_meetme.c, the relevant
options are stored in what are called conference flags and
Hi,
I have followed the article on how to install Asterisk with VM in IMAP but for
some reason it still continues to send it as a email. I have the following in
voicemail.conf :-
imapserver=
imapfolder=voicemail
imapport=143
expungeonhangup=yes
imapflags=notls
authuser=x
On Fri, 9 Oct 2009, Michelle Dupuis wrote:
Before we call each other liars and thieves, here is a link:
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
As with any open source, do your own homework on licensing, AND apply your
own reasonable judgment.
At this point I would
On Fri, 9 Oct 2009, Kevin P. Fleming wrote:
Gordon Henderson wrote:
While we're on the subject of G.729...
I can end to end use it with no transcoding, but voicemail is the main
sticking point for me - I'd need to transcode.
So why can't voicemail store the audio in the format it's being
On Fri, 2009-10-09 at 15:07 +0100, --[ UxBoD ]-- wrote:
Hi,
I have followed the article on how to install Asterisk with VM in IMAP but
for some reason it still continues to send it as a email. I have the
following in voicemail.conf :-
imapserver=
imapfolder=voicemail
Hello all,
I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
- Postpaid and prepaid applications.
True CDR,
Hello all,
I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
- Postpaid and prepaid applications.
- True CDR. Better that
A2billing (Star2Billing, I think, for commercial support) is a good
choice and it's very mature software.
Astercc is very fast and has a very good callshop solution.
Regards,
Juan
voip crazy wrote:
Hello all,
I want to instal a Billing solution in the same asterisk's box. I have
browse for
Gordon Henderson wrote:
All deskphnoes I've ever bought support g729 natively. They also all
support G711. The wholesale termination services I use all support g729
too. The only fly in the oinkment is local PSTN connections which are
obviously g711a. Now if voicemail would just blindly
,
recordingcheck|20091009-194302|1255102982.3126) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20091009-194302|1255102982.3126: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [...@macro-record-enable:5
-Original Message-
Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see in
the
CLI:
Extension '8085255935' in context 'default' from '808xxx' does
not
exist. Rejecting call on channel 0/1, span 1
On Fri, 9 Oct 2009, Kevin P. Fleming wrote:
Gordon Henderson wrote:
All deskphnoes I've ever bought support g729 natively. They also all
support G711. The wholesale termination services I use all support g729
too. The only fly in the oinkment is local PSTN connections which are
obviously
On Fri, 9 Oct 2009, Gordon Henderson wrote:
On Fri, 9 Oct 2009, Kevin P. Fleming wrote:
Gordon Henderson wrote:
All deskphnoes I've ever bought support g729 natively. They also all
support G711. The wholesale termination services I use all support g729
too. The only fly in the oinkment
are you using chan_local?
try disabling the hardware DTMF.
Sent using my wired Blueberry.
On 10/9/09, nik600 nik...@gmail.com wrote:
Dear all
i have a TE205P connected to an Asterisk 1.2.18.
Yes i know, the version is old but since now the system was stable and
i don't have the necessity
I would be very surprised if that were true. Your phones speak many
codecs, but they negotiate with asterisk on registration which one they
will be using. They don't switch codecs based on the remote channel
(which they don't even know about). Today, if your phones are negotiating
729 on
On Fri, 9 Oct 2009, Moises Silva wrote:
I would be very surprised if that were true. Your phones speak many
codecs, but they negotiate with asterisk on registration which one they
will be using. They don't switch codecs based on the remote channel
(which they don't even know about).
Well,
in the dialplan I now use Set(CDR(accountcode)=...) in the context of an
incoming call. It looks like this works for me.
So I can keep track of which account is receiving which call... and thus
separating them.
On Fri, 2009-10-09 at 13:19 +0200, Dovid Bender wrote:
I don't think
How can i activate ChanSpy to spy on a dedicated extension?
I see the following in /etc/asterisk/extensions_additional.conf
[chanspy]
include = chanspy-custom
exten = 501**,1,Chanspy(801)
exten = 501**,n,Hangup
exten = 502**,1,Chanspy(802)
exten = 502**,n,Hangup
But when i try to call 501**,
Use ExtenSpy for spying on a specific extension.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy
On 9 Oct, 2009, at 10:44 AM, Torintino T wrote:
How can i activate ChanSpy to spy on a dedicated extension?
I see the following in /etc/asterisk/extensions_additional.conf
Edited the post
How can i activate ChanSpy to spy on a dedicated extension?
I see the following in /etc/asterisk/extensions_additional.conf
[chanspy]
include = chanspy-custom
exten = 501**,1,Chanspy(501)
exten = 501**,n,Hangup
exten = 502**,1,Chanspy(502)
exten = 502**,n,Hangup
But
Landy Landy wrote:
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in
to one of those clone cards. Everything seem to be working fine. The only
problem I have is that I can't get voicemails coming from the PSTN line. All
other: SIP, IAX work fine. I can
On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote:
-Original Message-
Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see in
the
CLI:
Extension '8085255935' in context 'default' from
I think this is a dialplan problem. I would code it this way:
[chanspy]
include = chanspy-custom
exten = 5010,1,Chanspy(501)
exten = 5010,n,Hangup
exten = 5020,1,Chanspy(502)
Exten = 5020,n,Hangup
That way 5010 would spy on 501 and 5020 would spy on 502.
_
From:
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the
Sorry, I'm brand new at Asterisk (and/or FreePBX). I'm going to have to
figure out what all those things are before I can show them.
I'll have to get back to you.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
I believe it may be because you have not told what context the local channel
should use. Try using:
Channel: local/15...@mycontext
Obviously change the mycontext to the name of the context that you want to use.
That may work for you.
-Original Message-
From:
Hi
After a day of running asterisk, I got choppy sound when fw ip-pstn
When I restart asterisk the sound is fine,
Anyone had same problem?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 -
It would be helpful to know the OS, release of Asterisk, hardware, etc.
In my case, I start getting excessive echoes at end of day, so I do a
restart when convenient each morning around 4:00 AM.
_
From: asterisk-users-boun...@lists.digium.com
I've tried fourvariations on this theme:
Channel: local/15...@default
Channel: local/15...@dialout
Channel: local/1/default
Channel: local/1/Dialout
Neither one worked. I appreciate your time. Any other ideas?
Mike.
P.S. I thought that setting the context in the
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to
point out whatever I'm missing, no matter how stupid.
Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get:
Rejected connect attempt from 64.2.142.19, who was trying to reach
'6031234567@'
This leads
Hi,
I am using CentOS
Asterisk 1.4
The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware
echo cancelation
Does hardware echo will help?
Thanks.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
By the way, how to schedule auto reboot?
thanks
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, October 09, 2009 11:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Dear all,
According to:
http://www.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/
Citibankhas been under a telephone calling attack in 20th september.
Does anyone in asterisk community got any CDRs or logging of similar
attacks as the one above
Dear all,
According to:
w w w
.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/
Citibankhas been under a telephone calling attack in 20th september.
Does anyone in asterisk community got any CDRs or logging of similar
attacks as the one above
Hello guys,
I'm wondering what is required and involved in order to provide a
wifi/GSM handover to customers.
After googling I haven't found any product/vendor. Do you have an idea ?
Thanks in advance
Patrick
___
-- Bandwidth and Colocation Provided
I don't know if maybe you just sanitized your message for posting to
the list but the number coming in from vitelity is different from what
you've got in extensions.conf…also not seeing any of the necessary
peer definitions in your iax.conf sample to be able to accept the call
from
On Sat, Oct 10, 2009 at 03:15:20AM +0200, Patrick wrote:
Hello guys,
I'm wondering what is required and involved in order to provide a
wifi/GSM handover to customers.
After googling I haven't found any product/vendor. Do you have an idea ?
That's called UMA and you need operator cooperation.
Usually that message comes up because the caller is anonymous and
freepbx doesn't like anonymous calls by default.
There is an option to accept anonymous calls, or set the incoming trunk
to accept calls from the specific IP address
Of course it could be something else
Cheers Duncan
Ben
There are two commercial vendors that come to mind, namely DiVitas and
Agito.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick
Sent: Friday, October 09, 2009 8:15 PM
To: Asterisk Users Mailing
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