We upgraded to cvs-head a couple of days ago, but haven't changed the
config files at all.
Prior to the upgrade, all inbound calls to our call queues were
recorded, and the filename was like "agent-6043-1109793719-24472.gsm".
After the upgrade, the filename is now "1109793719-24472.gsm"
What
On Fri, 17 Jun 2005, Paul Redstone wrote:
> We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
> softphones and one hardphone.
>
> Working well except that we sometimes get dropped connections between IAX and
> the server with a max retries exceed message, which comes fro
We're using an SC420 and using BRI with a quadbri Junganns card, with IAX
softphones and one hardphone.
Working well except that we sometimes get dropped connections between IAX and
the server with a max retries exceed message, which comes from the chan_iax
driver code. The BRI side of things l
I'd like to be able to configure Asterisk to detect and respond to message
waiting indications from my telco, as I'm using the telco voicemail as a
fallback in case the Asterisk line is busy, Asterisk or it's server is down,
etc. Detecting either the FSK (preferred) or the stutter-tone, would allo
Hi guys !
Correct zaptel modules are probably missing, as wcfxo.
He must recopile zaptel on his Asterisk machine.
Best Regards,
Fracnois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sebastian
Silva
Envoyé : mercredi 15 juin 2005 21:
Hi all,
I got a small problem, my monitor is wacked :/ this is the result :(
No clue what to send with from logs/etc since ther eis no debug info
about it.
http://www.1av10.nu/~puppet/auto-1118970177-0850007222-1002.mp3
--
Daniel Eriksson
[EMAIL PROTECTED]
__
On Thu, 2005-06-16 at 23:24 -0400, Nate Kapi wrote:
When using the dial command and the D option to send DTMF digits when
the channel is answered, is there a way to allow for some dead air,
and then send more DTMF digits? I would like to automate a call, and
it requires entry of a few short d
When using the dial command and the D option to send DTMF digits when
the channel is answered, is there a way to allow for some dead air,
and then send more DTMF digits? I would like to automate a call, and
it requires entry of a few short dtmf digits all a couple seconds
apart from each other.
Th
Americo Sanchez C. wrote:
From: David John Walsh <[EMAIL PROTECTED]>
Reply-To: David John Walsh <[EMAIL PROTECTED]>,Asterisk Users
Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds
Date: Thu, 1
Hi all,
After upgrading to latest CVS head, I have problems using a IAXY device,
having slin problems:
Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping
incompatible voice frame on IAX2/lise-1 of format slin since our native
format has changed to ulaw
Because of that outside caller
I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3.
Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound ca
Conrad,
Heh, try asking about line appearances and the hint priority. People
clam right up. Or ask about receptionist phones that show all your line
statuses.
You can practically hear the crickets. :)
Sean
Conrad Beckert wrote:
To all who have answered my question: What a great mailing
It gets even better here in the US.
You can prepay for you cell service, but your unused mintues expire
after 90 days, and you forfeit your balance.
Greed is good.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle
Sent: Thursday, June 16,
>-Original Message-
>From: [EMAIL PROTECTED]
[mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of Terry H.
Gilsenan
>Sent: Thursday, June 16, 2005 6:24 PM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: RE: [Asterisk-Users] Bill seconds
> act, and why having ce
Nope, I'm not sure... I read it somewhere, but should have kept big mouth
shut! Can't find anything to back it up.
*attempts to remove foot from mouth*
I KNOW I read it... I'll look again later on to see if I can't find the
source.
*starts going back through his browser history on 5 different c
From: David John Walsh <[EMAIL PROTECTED]>
Reply-To: David John Walsh <[EMAIL PROTECTED]>,Asterisk Users
Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds
Date: Thu, 16 Jun 2005 18:36:00 +0100
An
From: "Leon Sun" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Subject: RE: [Asterisk-Users] Bill seconds
Date: Thu, 16 Jun 2005 10:56:23 -0700
The easiest way is to change another vend
Dan Littlejohn wrote:
If someone could point me to the incoming/outfoing call recording
feature for AMP it would be greatly appreciated.
Look in the Extensions Admin. Per Extension you can set:
Record INCOMING
Record OUTGOING
The script /var/lib/asterisk/bin/archive_recordings will create
They keep breaking the FAX support in libtiff. 3.6.1 is broken, although
a lot of distributions contain a patched version which works - usually
because a spandsp user got the patch pushed into the distribution.
HylaFAX users seemed to just give up trying to follow the buggy path of
libtiff, and
j_amorim wrote:
Hello guys,
I am having problems to installing libsupertone library.
The ./configure --prefix=/usr does not create the Makefile.
Any tip?
It has worked for everyone else. Lots of people have successfully
employed this library.
Steve
___
Terry H. Gilsenan wrote:
And as for cell phones being cheap, you have a receiver pays setup! How good
is that, then you have so many competing Telcos that sometimes you just
I believe that's unique to the US, the idea of paying for actually
receiving calls... don't know why they stand for it
You don't have to use queues to use agents. Do a show application dial
and look at what he is showing you.
You can have a macro run upon answer so put your menu there.
Kevin
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On 6/16/05, C F <[EMAIL PROTECTED]> wrote:
> This should compile against HEAD, this also includes a priority +101
> if the current parking spot is already in use.
Compiles fine, thanks much!!
Chris
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Jon,
thanks for your help, but I'd rather not do it using agents and queues,
ideally what would happen is it would simply play the message and wait
for the person to press a button, if nothing is pressed, it just keeps
going down the list. Any other suggestions?
[EMAIL PROTECTED]
wrote:
Da
enough said
--
Nicolás Gudiño
Buenos Aires - Argentina
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> Yes, meetme requires a clock source. You could try ztdummy. I tried
> using an FXO card as a clock source and observed that SIP calls connected
> to the conference seemed to get out of sync. Basically, after perhaps 20
> minutes or so in conference there was a 2 - 3 second delay between the
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of C F
> Sent: Friday, 17 June 2005 9:07 AM
> To: David John Walsh; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bill seconds
>
> On 6/16/05, David John Walsh
Yes, meetme requires a clock source. You could try ztdummy. I tried
using an FXO card as a clock source and observed that SIP calls connected
to the conference seemed to get out of sync. Basically, after perhaps 20
minutes or so in conference there was a 2 - 3 second delay between the
time that
Yes, Meetme needs timing. You can install ztdummy.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
You also need to recompile asterisk after you compile and install
zaptel.
Kevin
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Here is a link to some known issues...
http://www.soft-switch.org/spandsp-known-issues.html
-Original Message-
From: Roger Schreiter
Sent: Thu, June 16, 2005 6:53 pm
Hi,
I just started my very first attempt receiving
faxes by asterisk.
Compiling, installing and setup went without prob
I am grouping my extensions by building like so:
1XX is Building 1
2XX is Building 2
7XX is Office
[Office] extensions has the following includes 7xx
Include => Local
Include => International
Include => Building1
Include => Building2
[Building1] has
1xx
Include => Office
Include => Building
Hi Peoples
I am having problems with meetme, in that it responds
with “conf-invalid” when I dial a conference number.
I notice that there is a note with regards to
ztdummy, and the need for that to be loaded. Is this still the case?
Is meetme dependent on this module? I do NOT
This should compile against HEAD, this also includes a priority +101
if the current parking spot is already in use.
On 6/14/05, snacktime <[EMAIL PROTECTED]> wrote:
> I'm looking for a solution for call parking in an environment where
> multiple users are hosted on a single instance of asterisk.
I'm trying to make the fax detect to work, without luck
Regards,
--
Ing CIP Alejandro Celi Mariátegui
<[EMAIL PROTECTED]>
El jue, 16-06-2005 a las 13:35, Greg Blakely escribió:
> My Asterisk fax detection used to work, but no longer does.
>
>
> OK. So, here's the deal:
>
> 1. It appear
This is not an option for me, as the IVR menu is nuked as well...
Luki wrote:
A BETA firmware upgrade toasted my ATA286. It now has limited operations.
Happened to me too... looked mostly dead, but not quite.
Try a complete hardware reset. See section 8 on
http://www.grandstream.com/user_m
On 6/16/05, David John Walsh <[EMAIL PROTECTED]> wrote:
> Another way I have seen this done is to sell units, not pounds and pence
> credit
>
> eg a £2 calling card has 160 units (ratio of 80 units to the pound).
>
> If you were to charge 8p per min you make that 8 units per min. This
> gives
Hi,
I just started my very first attempt receiving
faxes by asterisk.
Compiling, installing and setup went without problems.
(asterisk-1.0.7, libtiff-3.6.1, SuSE-Linux 9.1)
When receiving a fax also everthings seems to work fine,
but the tiff file itsself is corrupted. Various tiff viewers
do r
If I have multiple Sipura 3000 device how can I dial out properly? I
can receive call without any problem and that's working really well.
Caller ID is shown and when someone call he get's the welcome message
the same way I have it configure with the X100P card. I don't seem to
have any echo
To all who have answered my question: What a great mailing list - not even 1
hour wait and yet such a lot of qualified answers!
Thank you very much
Conrad
Am Freitag, 17. Juni 2005 00:03 schrieb Sean Kennedy:
> Would see everyone else has already answered your question, so let me
> give you som
Would see everyone else has already answered your question, so let me
give you some background on it.
When * loads you can see ( if you do verbose that is ) all the modules
it's loading. Stock * loads more than I use, so I went through and
wrote down all the modules I wasn't going to use ( or tho
Right, turns out I am an idiot and I do have Asterisk running on 5070
instead of 5061. It's all working.
Now, if I could find out why calls coming from PSTN have horrible
voice quality
On 6/16/05, Luki <[EMAIL PROTECTED]> wrote:
> > I can see on tcpdump traces that the Invite packets
> > do g
There is a better way. Upgrade to the latest 3.1.3a firmware, which supports
PSTN to VoIP gateway calls. Specifically the "Off hook while calling VoIP"
option needs to be no.
If you need help configuring it, use the SPA3K w/Asterisk configurator at
Voxilla http://voxilla.com/spa3kasterisk.php
> Anyone know what I need to do to get the FXO port on the SPA 3000 to
> forward calls to Asterisk? My Asterisk is running on port 5061 and I
> set the dial plan on the device to forward to [EMAIL PROTECTED]:5061 but
> Asterisk is not picking it up. I can see on tcpdump traces that the
> Invite
> I can see on tcpdump traces that the Invite packets
> do go to through to the asterisk machine on port 5061,
> but it's not picking them up. sip debug does not show
> any packets either.
That would imply that the Sipura config is fine, but your Asterisk
setup is not listening at the right inter
On Fri, 2005-06-17 at 22:34 +0200, Conrad Beckert wrote:
> Hi
>
> ... probably one of those RTFM kind of questions (while I'd be happy to know
> where a good reference "FM" is :-) )
>
> Has anyone an idea on how to disable the console sound driver. My problem is
> that a running asterisk is mu
Hello guys,
I am having problems to installing libsupertone library.
The ./configure --prefix=/usr does not create the Makefile.
Any tip?
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On 20:55, Thu 16 Jun 05, Florian Overkamp wrote:
> Hi Michiel,
>
> > -Original Message-
> > Anyone who can help me with this ?
> > I tried everything :(
>
> > > exten => s,4,Dial(Local/[EMAIL PROTECTED],5,tTr)
> > > exten => s,5,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],10,tTr
modules.conf:
noload => chan_alsa.so
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 17, 2005, at 13:34, Conrad Beckert wrote:
Hi
... probably one of those RTFM kind of
Did you try with "noload => chan_alsa.so" and "noload => chan_oss.so" in
your modules.conf?
Sebas
Conrad Beckert wrote:
Hi
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference "FM" is :-) )
Has anyone an idea on how to disable the console sou
In /etc/asterisk/modules.conf
noload => chan_alsa.so
noload => chan_oss.so
- Dan
Conrad Beckert wrote:
>Hi
>
>... probably one of those RTFM kind of questions (while I'd be happy to know
>where a good reference "FM" is :-) )
>
>Has anyone an idea on how to disable the console sound driver. M
LiveJournal, a site with over a million users, uses Asterisk as the basic of their 'voice blog' system. On 6/16/05, Bill McLaughlin <
[EMAIL PROTECTED]> wrote:Hrmmm.. ummm... Not off hand. I read it on one of the VOIP sites( a news
article, not a forum-type post...)... I'll try to find a link.---
Likewise here, even looked through the confs cant see where it is
activated.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dan Littlejohn
> Sent: Thursday, 16 June 2005 4:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Dis
Hi
... probably one of those RTFM kind of questions (while I'd be happy to know
where a good reference "FM" is :-) )
Has anyone an idea on how to disable the console sound driver. My problem is
that a running asterisk is muting my speakers.
Thank you in advance for your help
Conrad
___
Bad form to post to your own mailing, but I found the flash panel docs
(http://www.asternic.org/) Oh well (I was blind or something)
If someone could point me to the incoming/outfoing call recording
feature for AMP it would be greatly appreciated.
Dan
On 6/16/05, Dan Littlejohn <[EMAIL PROTECTE
I have been looking through what I can find on @Home and AMP (wiki,
coalescentsystems.ca, maillists) and cannot find any documentation on
the incoming/outfoing call recording feature. If someone could some
point me to some I would be grateful. (and also for the flash panel,
like the default passw
I need help to make a conection form FWD to my pbx,
I can receive a call from PSTN for a FXo card but know I need to receive call
via IP form FWD I have activate hte IAX on freeworlddialup but does not work I
can't make or receive calls. I virtually new in this can please somebody help
me.
Hello,
I want to use an asterisk device as a load balancer. The idea is to
create a dialplan that would balance the incoming calls between X
different asterisk boxes (who will then handle the real job).
The most basic solution would be to do round-robin, the most elegant
would be to check fo
Hrmmm.. ummm... Not off hand. I read it on one of the VOIP sites( a news
article, not a forum-type post...)
... I'll try to find a link.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Thursday, June 16, 2005 12:18 PM
To: Asterisk
Anyone know what I need to do to get the FXO port on the SPA 3000 to
forward calls to Asterisk? My Asterisk is running on port 5061 and I
set the dial plan on the device to forward to [EMAIL PROTECTED]:5061 but
Asterisk is not picking it up. I can see on tcpdump traces that the
Invite packets do
On 6/16/05, Boris Bakchiev <[EMAIL PROTECTED]> wrote:
> He is using HA so I'm assuming he is running Master-Slave combo.
Yes this is correct, I am running Master-Slave. So you are saying
that iax2 should work just fine with virtual interfaces?
--
Thanks,
Lance Grover
__
I checked today : span 1 is *definately* connected to the pstn, as is
span 3. spans 2 and 4 are connected to the meridian.
my sync is on span 1 (primary) and 3 (secondary) as defined in the
zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48
Has anyone gotten this tester to work? i can get it to log in and
show me my call load.. but it doesn't seem to MAKE any calls.
On 3/30/05, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
> Bicom Systems wrote:
> > [EMAIL PROTECTED] wrote:
> >
> >>Its a very very bad idea to do this on production
- Original Message -
From: "Jim Duda" <[EMAIL PROTECTED]>
Jun 13 01:04:47 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
I guess I can write a perl script which uses the API command interface
to poll and attempt a re registry every so often ... I haven't looked
at
We are getting HDLC errors on a PRI with a Dell PowerEdge SC420. I
suspect it may be an interrupt issue.
Can anyone recommend a low cost name brand server that will not share
the interrupts or have the issues that the Dell PowerEdge SC420.
Thanks
__
Hi Michiel,
> -Original Message-
> Anyone who can help me with this ?
> I tried everything :(
> > exten => s,4,Dial(Local/[EMAIL PROTECTED],5,tTr)
> > exten => s,5,Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED],10,tTr)
Have you tried using the /n parameter for chan_local ? I've no
I have these messages in my log file
Jun 12 23:33:57 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun 12 23:34:47 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun 13 00:19:00 NOTICE[1707]: Registration of 'jduda' rejected:
Registration Refused
Jun 13
> I use the Teliax service with the IAX2 protocol. I noticed 2 days ago
> that I was not registered with the Teliax server. I used the "iax2 show
> registry" command and found I was not registered with Teliax. I issued
> a reload command in asterisk in order to connect again.
>
> I went to the
My Asterisk fax detection used to work, but no longer does.
OK. So, here's the deal:
1. It appears that the "faxdetect" command cannot be applied
channel-by-channel in zapata.conf anymore, as Asterisk appears to the
last "faxdetect=" command to ALL channels.
2. My stations are detected and s
Are you sure about that?
I know Freshtel.net uses a highly customized version of asterisk.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Bill McLaughlin
> Sent: Thursday, 16 June 2005 2:12 PM
> To: [EMAIL PROTECTED]; 'Asterisk Use
Bill McLaughlin wrote:
Vonage uses Asterisk, and they have a lot more than 3000 customers.
??
You have documentation of that assertion?
Not saying you're wrong, but I've never seen such a thing before.
B.
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Asteris
On 6/16/05, Bill McLaughlin <[EMAIL PROTECTED]> wrote:
> Vonage uses Asterisk, and they have a lot more than 3000 customers.
That should help your argument!
Michael
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Are you using, putting those lines in the mgcp.conf file, should handle two
lines?
Did anybody tried it?
Thanks
Quoting Florian Overkamp <[EMAIL PROTECTED]>:
> Hi,
>
> > -Original Message-
> > Anybody here know or using Asterisk with 2 lines MGCP phone?
> > I am trying to
> > figure
I use the Teliax service with the IAX2 protocol. I noticed 2 days ago
that I was not registered with the Teliax server. I used the "iax2 show
registry" command and found I was not registered with Teliax. I issued
a reload command in asterisk in order to connect again.
I went to the Teliax websi
Vonage uses Asterisk, and they have a lot more than 3000 customers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Stearne
Sent: Thursday, June 16, 2005 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Us
Anyone who can help me with this ?
I tried everything :(
On 14:26, Tue 14 Jun 05, Michiel van Baak wrote:
> Hi list,
>
> For months everything worked super here in our setup.
> This week I implemented some new idea in our webbased
> calendar system. I thought it would be nice to have an
> option
On Thu, 16 Jun 2005, Rich Adamson wrote:
>
> The E1 card does not receive "clocking" from any span. It "sync's"
> the on-board clock to whatever span you choose. If you watch what
> others have posted on the list over many months, you'll notice many
> have never specified a clock sync source. T
Hi,
> -Original Message-
> Anybody here know or using Asterisk with 2 lines MGCP phone?
> I am trying to
> figure out if there are such device available and if so, how does it
> differenciate between the lines that is associated with
> extention number.
Theoretically you could differe
If you need a SIP 30+6 a-z carrier, let me know. We may do 6+6 for you.
Leon Sun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: June 15, 2005 10:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-U
The easiest way is to change another vendor asap. It is ridiculous that your
carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter and
billing unit does.
Leon Sun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: June
On 6/16/05, Alistair Cunningham <[EMAIL PROTECTED]> wrote:
> Several people have responded with architecture suggestions. While these
> are welcome, I'm happy with the architecture options planned, having
> done many large voicemail implementations on products other than Asterisk.
>
> What I had h
Hi,
I currently have a hardware PBX with its own custom phones, which I'm using
pretty much as an in-house intercom system. At some point in the future I
might want to convert to Asterisk. My setup is two analog POTS lines from
my local phone company and currently nine stations or extensions, eac
Another way I have seen this done is to sell units, not pounds and pence credit
eg a £2 calling card has 160 units (ratio of 80 units to the pound).
If you were to charge 8p per min you make that 8 units per min. This
gives you a 20% increase which might help if your on per second
billing to yo
Hi all,
I am using ASTCC
From: Darren Wiebe <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bill seconds
Date: Wed, 15 Jun 2005 23:05:54 -0600
I've done a little
Hi list!
I have an asterisk box connected to an ADSL connection that has 1 Mbit
upstream. Is there any way to use max channels intelligently?
For example I would like to do some checks on the outgoing calls. When
it's quiet I want each and every call to go out to my IAX provider.
However wh
hi, i am using iax. i am setting up a new asterisk box
#2 on my network. It is "behind" another asterisk
box#1. Box#1 acts as a
router/firewall/asterisk/nat/dhcp. It has a public IP
on ethernetcard1 and a private ip on ethernetcard2.
box #2 has a private ip.
I have a DID from teliax. When I c
On Thu, Jun 16, 2005 at 10:45:04AM -0600, Tore Hansen said:
> I am interested in creating a telemarketing call blocker in my Asterisk
> dial plan. I am not much of a programmer, and I am wondering if external
> AGI code would be required to implement this.
>
> The logic that I would like to have
Hi,
I am interested in creating a telemarketing call blocker in my Asterisk
dial plan. I am not much of a programmer, and I am wondering if external
AGI code would be required to implement this.
The logic that I would like to have in place is this:
1. If the incoming call carries proper name
Several people have responded with architecture suggestions. While these
are welcome, I'm happy with the architecture options planned, having
done many large voicemail implementations on products other than Asterisk.
What I had hoped to get from Asterisk-Users and Asterisk-Biz was not a
techni
Michael,
Yes, this is exactly what we plan to do.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Michael Stearne wrote:
On 6/16/05, Alistair Cunningham <[EMAIL PROTECTED]> wrote:
I'm planning an Asterisk Voicemail system of around 3000 users spread
across seve
William,
I'm happy with the architecture options, as this is a company WAN with
dedicated fibre links, and I'll be securing the database and NFS servers
comprehensively.
All I need are case studies to assure the customer that they're not the
first people to do this on Asterisk. Positive repo
On Jun 16, 2005, at 8:23 AM, Kumara Jayaweera wrote:
Starting simple switch on 'Zap/1-1'
-- Executing
Dial("Zap/1-1","IAX2/[EMAIL PROTECTED]/10094472239112|30|tr")
in new stack
-- Called [EMAIL PROTECTED]/10094472239112
What country code is that you're dialing?
Robert Goodyear
I posted this http://lists.digium.com/pipermail/asterisk-users/2005-
June/111815.html and never received a response. I just wanted to
share with you that I think I fixed the problem. The only thing I
changed was my Dial command by removing the 'r' option. Since then,
asterisk seems to proper
On 6/16/05, Alistair Cunningham <[EMAIL PROTECTED]> wrote:
> I'm planning an Asterisk Voicemail system of around 3000 users spread
> across several sites, each site connected by a fast network to a central
> site. We're considering 2 models:
>
> - Central Voicemail with VoIP calls from remote site
Hi,
we test the misdn module together with beronet BN8S0 card.
We connect the pstn ISDN line to Port 1 and an ISDN phone to Port 2.
That works great, the ISDN phone rings an we can make the call.
When the caller hangsup before call is answered by the callee the call
on Port 2 rings until en
On Thu, Jun 16, 2005 at 03:27:49PM +0100, Alistair Cunningham wrote:
> I'm planning an Asterisk Voicemail system of around 3000 users spread
> across several sites, each site connected by a fast network to a central
> site. We're considering 2 models:
>
> - Central Voicemail with VoIP calls from
HI,
Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to
figure out if there are such device available and if so, how does it
differenciate between the lines that is associated with extention number.
Thanks
___
Asterisk-Users ma
Scrap this question.. found the answer later... so I'm using ODBC...
but for some reason varchar(80) is coming in as 80 characters.. if say
CLID is only 10 characters it will appear as "5703332121
[80 characters] "
any ideas?
On 6/16/05, Matt <[EMAIL PROTECTED]> w
[EMAIL PROTECTED] wrote:
> Actually what happens is that from SER debug I can see the call is
> looping between Asterisk and SER. but adding a number makes no
> loops.
Check what the origin (IP/DNS name) of the incoming SIP message is.
If it's from asterisk, send it to the user, if it is not fr
I soo want this feature. This would be the last hurdle in getting
off my Lucent/Avaya Definity G3.
Mark
Tim Connolly wrote:
Has anyone figured out how to mimick a traditional bridged-appearance? My
guys like the ability to put a call on hold on line "3" and it's the same
call on line "3"
I'm trying to use freetds/odbc to write CDR records to a MSSQL
database but when I installed them and tried to compile asterisk again
I get:
_tds.c
cdr_tds.c: In function `mssql_connect':
cdr_tds.c:415: `TDSCONNECTINFO' undeclared (first use in this function)
cdr_tds.c:415: (Each undeclared identi
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