My asterisk is giving me problems when I use it as a pstn gateway to SER ,
basically what happens is that its either I get one way audio or no audio
at all when I make pstn calls via asterisk from sip clients registered
with SER.
___
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Hi Users,I'm new to Asterisk programming , I'm in working the Voip Technologies by using the OpenSER for my call routing process and Radius For AAA.But in Asterisk i need it for only PBX and VoiceMail,For Account I'm using the Openser + Radius .
Main My doubt is that, For Call Routing my using the
If you do VoiceMail(bEXTEN) then all you get is Please leave your
message after the tone so you can do the greeting function in the
dialplan.
If you do VoiceMail(bEXTEN) then you get name is on the phone.
Please leave...
(uEXTEN) gets you name is unavailable. Please leave...
On 9/15/06,
I followed my intuition and sent a DTMF tone via the D() option in the
dial command to the destination and that causes the voice to be
transmitted after the call is answered.
I also realised that the ringing tone to the destination does not come
up until I execute a SendDTMF command before
Hi list,
is it possible to call a shell script from * which returns a number or a
string which can be read to an asterisk variable? Something like
'Set(VAR(System(/opt/scripts/something.script)))?
Does anyone have an idea?
Regards, Christophorus
___
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:
I have been searching, but I have not found the answer.. How might I add
the amr codec to my asterisk server?
I believe I found the amr source from
http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip
I compiled it but
Currently, I send a popup to an agent asking them to accept the call or
not. However, they are complaining that they do not *see* the popup
(it's onscreen for 7 seconds) and I was wanting (have been asked to
provide) a facility where I can ping an agents phone before sending
the popup.
This
Matt,
I am sure this is a RTFM and I am pretty sure you are using meetme
rooms. Just not too sure how you do the magic.
28 T1s with NFAS so 95 channels per trunk group, seven trunk groups =
665 lines. My client's call volume has shot from 5,000 to about 10,000
calls a day. Due to recent
How large is the popup?
I cant imagine when my outlook 2003 email popup appears (5 seconds I
guess) that I don't see it, though I think the movement of it appearing
on my display then receeding away has a lot to do with it.
Cheers,
Dean
-Original Message-
From: [EMAIL
Um, it's a 1024x768 bright yellow box ;)
I know, unbelievable.
Julian.
Dean Collins wrote:
How large is the popup?
I cant imagine when my outlook 2003 email popup appears (5 seconds I
guess) that I don't see it, though I think the movement of it appearing
on my display then receeding away
Julian Lyndon-Smith wrote:
Currently, I send a popup to an agent asking them to accept the call
or not. However, they are complaining that they do not *see* the popup
(it's onscreen for 7 seconds) and I was wanting (have been asked to
provide) a facility where I can ping an agents phone before
hi all
Linked from /. today, http://www.networkworld.com/news/2006/091206-
von-sam-houston.html talks about the Sam Houston State University
(SHSU) migrating a rather large amount of users to asterisk. The
article describes the installation in rather vague terms, so I was
wondering if
Roy Sigurd Karlsbakk wrote:
hi all
Linked from /. today,
http://www.networkworld.com/news/2006/091206-von-sam-houston.html
talks about the Sam Houston State University (SHSU) migrating a
rather large amount of users to asterisk. The article describes the
installation in rather vague terms,
Hello,
At this point in time VICIDIAL is more focused on outbound features,
but inbound and blended capcbilities have been part of VICIDIAL for
about two years. The most I have done inbound-only with it is 3 T1s
with 60 agents. But for outbound and inbound agents together we have
had upto 120
Right now we are all inbound and every call is recorded.
Matt Florell wrote:
Hello,
At this point in time VICIDIAL is more focused on outbound features,
but inbound and blended capcbilities have been part of VICIDIAL for
about two years. The most I have done inbound-only with it is 3 T1s
with
Linked from /. today, http://www.networkworld.com/news/2006/091206-
von-sam-houston.html talks about the Sam Houston State University
(SHSU) migrating a rather large amount of users to asterisk. The
article describes the installation in rather vague terms, so I was
wondering if someone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Roy Sigurd Karlsbakk wrote:
Linked from /. today,
http://www.networkworld.com/news/2006/091206-von-sam-houston.html
talks about the Sam Houston State University (SHSU) migrating a
rather large amount of users to asterisk. The article describes
Maybe I should just hold a conference call about all this stuff.
On Sat, 2006-09-16 at 18:21 +0200, Matt Riddell (IT) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Roy Sigurd Karlsbakk wrote:
Linked from /. today,
http://www.networkworld.com/news/2006/091206-von-sam-houston.html
Guys,
this may be a off question but where do I start to find all the
regulations to comply with to set up a residential voip phone service
in the US?
The tech stuff I am fine with but the legal is where I need some
pointers.
Also I am not looking for a flame war just pointers on the legal
To maintain high recording quality with no audio skips we have found
that you should not go over 50 conversations being recorded on a
single server. What have you found is your limit while maintaining
very good audio quality?
MATT---
On 9/16/06, Steve Totaro [EMAIL PROTECTED] wrote:
Right now
That would be superb!
-Original Message-
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat, 16 Sep 2006 11:32:00 -0500
Delivered: Sat, 16 Sep 2006 13:21:48
Siqhamo Sifo wrote:
My asterisk is giving me problems when I use it as a pstn gateway to SER ,
basically what happens is that its either I get one way audio or no audio
at all when I make pstn calls via asterisk from sip clients registered
with SER.
SER itself is just a SIP Proxy. So your
Precisely our configuration. Dell 1850 with 4 port PRI digium cards.
No issues on my last two consulting jobs.
/edg
--On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon
[EMAIL PROTECTED] wrote:
Hi, Alan,
We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards
in it
This information seems to indicate there is a problem with the 1850 and the
onboard nic.
http://connection-telecom.com/support.html
-Original Message-
From: Ed Greenberg [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 16, 2006 1:45 PM
To: Asterisk Users Mailing List -
On Fri, Sep 15, 2006 at 02:58:02PM +0200, Robert Rozman wrote:
I'm banging my head on compiling bristuff modules for Suse 10.0 with kernel
:
Linux laps1 2.6.13-15.11-smp #1 SMP Mon Jul 17 09:43:01 UTC 2006 x86_64
x86_64 x86_64 GNU/Linux
and Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s.
I
-Original Message-
From: Andrew Joakimsen [mailto:[EMAIL PROTECTED]
Sent: Friday, September 15, 2006 4:37 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: Re: [asterisk-users] Reliability of the newer IAXy's
This goes for everyone;
On Fri, Sep 15, 2006 at 11:09:51AM -0500, Juan Miguel Yamakawa wrote:
Help me please..
ZT_SPANCONFIG failed on span 1: No such device or address (6)
how can i fixed this problem.
This means that a span linein /etc/zaptel.conf did not fit the spans
that exist on your system now (see
yes in did, this could be one excellent case study for all asterisk Community! Please let me know when it will be held!
By the way, keep asterisk community following your steps would be great for knowledge of everyone and to solve possible problems you could find!
On 9/16/06, Melcon Moraes [EMAIL
On Sat, Sep 16, 2006 at 01:15:01PM +0200, Christophorus Laube wrote:
Hi list,
is it possible to call a shell script from * which returns a number or a
string which can be read to an asterisk variable? Something like
'Set(VAR(System(/opt/scripts/something.script)))?
Does anyone have an
So with that said, can anyone recommend a way that I can get a sip
client on a cell phone that uses H.263 and amr to talk to an asterisk
system?
Is it just not possible because of licensing? It sounds kind of lame to
have a sip client that can't talk to anything else because of codecs..
Steve
Yeh well I don't and I have just as much right to this list as you do.
I think it's fine for a company, commercial or otherwise to post product
announcements in the non -biz list.
As long as they only post it once, and it's not an announcement about a
price change or something equally lame (like
Would anyone be kind enough to post a sip.conf fragment as a sample for
use with a Mediatrix 1204?
___
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Bill Michaelson wrote:
Would anyone be kind enough to post a sip.conf fragment as a sample for
use with a Mediatrix 1204?
Ours works with:
[mtrix1]
type=peer
host=172.28.4.46
mask=255.255.255.255
context=in-mtrix1
qualify=no
canreinvite=no
dtmfmode=inband
disallow=all
allow=ulaw
Best
Hi list, Ihave a asterisk and sopthonesworking well, and I make a configuration for it work with a voice gatewayAddpac 2120 (4port FXO y 4 ports FXS), I have connected my gateway to my PBX, when Itry to call to PSTN from my softphone, I have a trouble that Asterisk add the number 9overthe number
In your extensions.conf you probably have:
exten = _9nxxnxx,1,Dial(ZAP/g1/${EXTEN},180,r)
Change that to:
exten = _9nxxnxx,1,Dial(ZAP/g1/${EXTEN:1},180,r)
Of couse use the values that you need, not that exact example...
On 9/16/06, Pablo Almido [EMAIL PROTECTED] wrote:
Hi list, I
On 16 Sep 2006, at 20:38, Net Nut wrote:
So with that said, can anyone recommend a way that I can get a sip
client on a cell phone that uses H.263 and amr to talk to an asterisk
system?
Is it just not possible because of licensing? It sounds kind of
lame to
have a sip client that can't talk
Hello list!Is there a way to program one of the buttons on the 501 (Like the services button) to do on the fly call recording? So in the middle of the phonecall you can record the call without have to do a transfer type of setup. Ive looked at the manual but cant seem how to do that, I only see
So the config is stored in one db shared between two asterisk? What
you need to do is setup an IAX trunk between the two machines and then
tell it to dial UA2 on the local machine, and then the extension for
UA2 on the 2nd asterisk. Setup both the same on the other machine.
Now do you have the
Mauricio Mantilla mauriciomantilla at gmail.com writes:
Hi,I'm trying to connect my cell phone (motorola V3) to asterisk, using this
guide:
http://www.thetechguide.com/howto/asterisk/chanbluetooth.html
Everything has worked ok, but when I actually want to start
asterisk, my phone doesn't
On 2006-09-15 13:42:21 -0700, Lists [EMAIL PROTECTED] said:
Not sure what to tell you. But for the price, I might have to try one
of these instead:
And I for one would like a transcript or recording of it!
PaulH
Melcon Moraes wrote:
That would be superb!
-Original Message-
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Sent: Sat,
Hi, Actually I have several AGI scripts, manager Originates and Call
files doing different stuff applying the same procedure, thats why I
asked about console messages. It seems very weird to me.
Regards
On 9/16/06, Frank Church [EMAIL PROTECTED] wrote:
I followed my intuition and sent a DTMF
Hi,
I have just compiled
trion*CLI show version
Asterisk 1.2.12.1 built by root @ trion on a i686 running Linux on
2006-09-16 16:39:13 UTC
I have copyed the old configuration files from the stable debian
version 1:1.0.7.df
I have changed in sip.conf the bindport (port=) to (bindport=) .
I have the same setup as Florian, however I have dtmfmode set to rfc
instead of inband
On 9/16/06, Florian Overkamp [EMAIL PROTECTED] wrote:
Bill Michaelson wrote:
Would anyone be kind enough to post a sip.conf fragment as a sample for
use with a Mediatrix 1204?
Ours works with:
[mtrix1]
Steve Kennedy wrote:
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote:
I have been searching, but I have not found the answer.. How might I add
the amr codec to my asterisk server?
I believe I found the amr source from
On 9/17/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
So the config is stored in one db shared between two asterisk?
Yes, the configuration files are shared between them.
What you need to do is setup an IAX trunk between the two machines and then
tell it to dial UA2 on the local machine, and
How can I use a system cmd to get back the return value in dial plan?
Say, I want to run a script using system cmd to get the hostname.
System(hostname)
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