Hi lists,
Does anyone know if the following error message (from a debug screen) was
a
deliberate change from the behavior in asterisk V1.4.18 or just an
overlooked
parsing error in progressing to V1.6.0? Since, in this case, the string ("Hi
there")
is quoted, it doesn't seem as though the par
Hi Jaap,
Here is a portion of my coding. Hope I didn't 'break' anything in moving
sections around.
I tried to leave only what is material to answering your question.
One caveat - I did add the CALL FILE functionality on my V1.6.0 (branch
SVN)
asterisk box; so, maybe they changed some 'unde
Anyone seen anything on the IP670 & the Color Expansion?
-Matt
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Does anyone know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
differe and how these different servers will interact as a single logi
Quoting Michael Graves <[EMAIL PROTECTED]>:
>> in case anyone is interested, I've just taken ownership of a small home
>> network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone.
>>
>> It works great with Asterisk. ...
Sounds great, especially where you say that you got MWI to work wit
Try the RPM from Trixbox. If you need something to open the file on Windows,
7zip works fine..
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/firmware-polycom-0-3.0.1-2.html
-Jon
- Original Message -
From: "Darrick Hartman (lists)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing
On Monday 28 April 2008 17:30, Robert McNaught wrote:
> I am trying to write a custom application which will integrate with an
> existing MSSQL crm system.
>
> We need to get ahold of the CDR(uniqueid) field in during call-time -
> I see from doing a DumpChan(), the CDR unique ID is available as so
Hi,
I am trying to write a custom application which will integrate with an
existing MSSQL crm system.
We need to get ahold of the CDR(uniqueid) field in during call-time -
I see from doing a DumpChan(), the CDR unique ID is available as soon
as the call is created. CDRs usind odbc are only writt
On Mon, 28 Apr 2008 18:05:47 -0400, Andreas van dem Helge wrote:
>AFAIK Siemens ceased distribution of their Gigaset line in North
>America a few years ago either you find a wholesaler that is importing
>"grey market" items or you buy it from a distributor overseas.
Was there any particular reaso
Andreas van dem Helge wrote:
> Anyone have a download link for 3.0 SIP firmware?
>
> If you are going to say "ask polycom" or "ask your vendor" don't even
> waste your time posting. I've asked the Nazis and they'll probably
> take > 1 week.
Suggest you get a different vendor then. I got a respon
AFAIK Siemens ceased distribution of their Gigaset line in North
America a few years ago either you find a wholesaler that is importing
"grey market" items or you buy it from a distributor overseas.
On Sun, Apr 27, 2008 at 11:16 AM, Michael Graves <[EMAIL PROTECTED]> wrote:
> On Sun, 27 Apr 2008 1
Anyone have a download link for 3.0 SIP firmware?
If you are going to say "ask polycom" or "ask your vendor" don't even
waste your time posting. I've asked the Nazis and they'll probably
take > 1 week.
Thanks,
Andy
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Jerry Geis wrote:
> I have xinet tftp running on centos 5.1
>
> It seems to be running on the local network eht0 fine. My box has 2 nics.
> however when I connect to eth1 for tftp I get:
>
> in.tftpd[5084]: tftpd: read(ack): Connection refused
>
> How can I get tftp working on BOTH eth0 and eth1 f
On 4/28/08, Jay R. Ashworth <[EMAIL PROTECTED]> wrote:
> On Sat, Apr 26, 2008 at 11:15:42AM +1000, Rob Hillis wrote:
> >Two dual core processors would should four processors - each processor
> has
> >two virtual processors for a total of four.
>
> I *think* Rob wrote that; *please*, peo
I am going on memory but I do recall that Aastra had a phone that used
ADSI codes that would 'turn on' a speaker on an analog phone
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Monday, April 28, 2008 3:10 PM
>
To help you adjust your rxgain and txgain appropriately, you can ask
your telco for the phone number for a "milliwatt test line". 10 is a
pretty high number for the gain, although it DOES depend on your
distance from the telco and the line quality. My rxgain ranges between
2.375 and 2.945 dep
Hi men,
What happens after restarted xinetd ?
Only one Eth access again or suddently the two ?
Francois
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Jerry Geis
Envoyé : lundi 28 avril 2008 20:33
A : asterisk-users@lists.digium.com
Objet : Re: [asteris
>
> Looks good to me. Try this: after doing the service iptables stop do
> the following and see if there are any rules left:
>
> iptables -L
> iptables -t nat -L
> iptables -t mangle -L
>
> if there are any rules at all listed, replace the -L with -F and re-run
> the commands.
>
> -Brent
Ran
Jerry Geis wrote:
>
> Brent, below is the file. Looks good to me... Also Both networks start
> at boot. Nothing is manual on this box at all.
>
> --
>
> # Simple configuration file for xinetd
> #
> # Some defaults, and include /etc/xinetd.d/
>
> defaults
> {
>
Bilal,
So you want to page through your analog phones, no overhead paging. I
doubt this is possible.
I think your options are an IP phones such as the Polycom that
supports paging or using an AMP and speakers usually mounted high on
the wall or flush with a tile ceiling.
Thanks,
Steve Totaro
O
On Mon, Apr 28, 2008 at 12:48 PM, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> Makes me feel like the
> glassmaker sending the kid to break the window)
>
> --
>
>Tzafrir Cohen
> icq#16849755 jabber:[EMAIL PROTECTED]
> +972-50-7952406 mailto:[EMAIL PROTECTED
Assuming I wanted to
Dial 1 outside number, once connected I wanted to dial a second number and
bridge them. Would I do something to this effect?
[bridge]
;;
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(/sounds/pl
>
> Check your /etc/xinetd.conf file and see if the bind= line is blank or
> contains only one interface by some chance. Blank should cause xinetd
> to bind to all interfaces. Also, do you bring up eth1 at startup or do
> you bring it up manually after boot? If it's brought up after boot,
>
Jerry Geis wrote:
> The netstat show 0.0.0.0
>
> netstat -anp | grep :69
> udp0 0 0.0.0.0:69
> 0.0.0.0:* 4007/xinetd
>
> --
> cat /etc/xinetd.d/tftp
> # default: off
> # description: The tftp server serves files u
Hi,
Since a week ago I am trying to get chan_misdn working with asterisk
1.4.19, using HFC based ISDN card on Linux 2.6.22.
My setup is done as detailed on wiki and FAQ.
* mISDN and miSDNusers are 1.1.7.2, unpacked, build and installed.
After installation and misdn-init, I have this:
aragorn:
How can I get a list of the callers within a specific
queue at any given moment?
I need to get the caller IDs of all active calls in a
queue then send them out via a udp socket to a
listening application on the network (the only data I
need to send are two fields: current timestamp and
caller id o
Eve-Ellen Cole wrote:
> I am thinking of going with a Dell PowerEdge 1950 ||| for a new
> CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB
> memory, two 500GB hard drives (presumably mirrored). I also plan to get
> a Digium TE220B to go with it. (a non-dell server is not an
Jaap Winius wrote:
> Quoting Jerry Harshany <[EMAIL PROTECTED]>:
>
>
>> There is an additional alternative for a ringback to a caller, which
>> is to use the Call File capability as noted in Van Meggelen's
>> "Future of Telephone"; 2nd ed, p306.
>>
>
> As it says in the book, call fil
On Sunday 27 April 2008 12:19, Steve Totaro wrote:
> On Sun, Apr 27, 2008 at 12:42 PM, Tilghman Lesher
>
> <[EMAIL PROTECTED]> wrote:
> > On Sunday 27 April 2008 11:11:28 Steve Totaro wrote:
> > > On Sun, Apr 27, 2008 at 12:05 PM, Tilghman Lesher
> > >
> > > <[EMAIL PROTECTED]> wrote:
> > > > O
On Mon, Apr 28, 2008 at 10:58:14AM -0400, Steve Totaro wrote:
> Hence the reason I kept CCing your off-list emails back to the list.
> Guys like Tzafrir are aces.
(Just a reminder that if this is indeed the case then this is a bug
inflicted by me and fixed later by sruffell. Makes me feel like th
-
> > In the process of cleaning up unnecesary
> processes, I
> > came across this line :
> >
> > /usr/sbin/vmware-guestd --background
> > /var/run/vmware-guestd.pid
> >
> > GASP so does this mean this is a virtual
> machine??
> > I have got no idea about virtualization yet. So
> how do
> Try having a look at the settings by running 'lokkit' or
> 'system-config-security-level-tui' from the command lin - ensure that
> the firewall is disabled from there also, and turn off SELinux and see
> if that makes any difference.
>
> Robert
Robert,
I have turned off "service iptables stop"
The netstat show 0.0.0.0
netstat -anp | grep :69
udp0 0 0.0.0.0:69
0.0.0.0:* 4007/xinetd
--
cat /etc/xinetd.d/tftp
# default: off
# description: The tftp server serves files using the trivial file
transfer \
#
Try having a look at the settings by running 'lokkit' or
'system-config-security-level-tui' from the command lin - ensure that
the firewall is disabled from there also, and turn off SELinux and see
if that makes any difference.
Robert
On Mon, Apr 28, 2008 at 9:11 AM, Jerry Geis <[EMAIL PROTECTED]
Jerry Geis wrote:
> I have xinet tftp running on centos 5.1
>
> It seems to be running on the local network eht0 fine. My box has 2 nics.
> however when I connect to eth1 for tftp I get:
>
> in.tftpd[5084]: tftpd: read(ack): Connection refused
>
> How can I get tftp working on BOTH eth0 and eth
Xinetd may have bound the service to a particular IP address. Look at
your Xinetd.d config.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Monday, April 28, 2008 12:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asteris
On Sat, Apr 26, 2008 at 11:15:42AM +1000, Rob Hillis wrote:
>Two dual core processors would should four processors - each processor has
>two virtual processors for a total of four.
I *think* Rob wrote that; *please*, people, turn your HTML off on
mailing lists? :-)
Two dual-cores don't h
On Fri, Apr 25, 2008 at 12:29:08PM -0400, Steve Totaro wrote:
> My dual proc, dual core AMD boxen show as four procs. I guess the AMD
> architecture uses Hypertheading (or whatever the equivalent is for
> AMD, I assume Intel owns the rights to the name Hyperthreading).
No, those really *are* 4-pr
On Mon, Apr 28, 2008 at 11:07 AM, Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote:
>/ I have xinet tftp running on centos 5.1
/>/
/>/ It seems to be running on the local network eht0 fine. My box has 2 nics.
/>/ however when I connect to eth1 for tftp I get:
/>/
/>/
> But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2.
> Or is C2D not four cores?
If I'm not mistaken, there was a "Core Duo" - which was dual core processor.
Now there is a "Core 2 Duo" - which is a "second generation" dual core
processor.
Still just 2 cores though... (which
On Mon, Apr 28, 2008 at 11:07 AM, Jerry Geis <[EMAIL PROTECTED]> wrote:
> I have xinet tftp running on centos 5.1
>
> It seems to be running on the local network eht0 fine. My box has 2 nics.
> however when I connect to eth1 for tftp I get:
>
> in.tftpd[5084]: tftpd: read(ack): Connection refus
Quoting Jerry Harshany <[EMAIL PROTECTED]>:
> There is an additional alternative for a ringback to a caller, which
> is to use the Call File capability as noted in Van Meggelen's
> "Future of Telephone"; 2nd ed, p306.
As it says in the book, call files allow calls to be created through
th
I have xinet tftp running on centos 5.1
It seems to be running on the local network eht0 fine. My box has 2 nics.
however when I connect to eth1 for tftp I get:
in.tftpd[5084]: tftpd: read(ack): Connection refused
How can I get tftp working on BOTH eth0 and eth1 for my phone config files.
man
Eve-Ellen,
I hear you on the buying now rather than fighting later :)
Maybe LDAP can help with the import. Anyways, there are pretty simple
ways around that providing you can export what you have as a CSV or
you could set something up with realtime (ughh, I hate realtime but it
may be of benefit
On Mon, Apr 28, 2008 at 10:31 AM, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
> On Mon, Apr 28, 2008 at 09:30:18AM -0400, Steve Totaro wrote:
> > On Mon, Apr 28, 2008 at 9:20 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> > > On Mon, Apr 28, 2008 at 03:07:46PM +0200, Philipp Kempgen wrote:
> > >
>
On Mon, Apr 28, 2008 at 10:27 AM, Arthur <[EMAIL PROTECTED]> wrote:
> >
> > I can dial out to other numbers without issue.
> >
> >
> > Calling the number from a separate PSTN phone works fine.
>
>
> I have had such problem last week. from PRI interface (i get busy tone as
> fast as i finish ty
On Mon, Apr 28, 2008 at 10:28 AM, John Faubion <[EMAIL PROTECTED]> wrote:
> > > need auto answer or how else would the audio come out of
> > the speakers?
>
> One of the lowest cost solutions I've used for this is a Grandstream BT-200.
> You can buy the phone for around $50, it supports auto ans
Ah, managing user accounts. That is going to be very challenging. I
haven't looked at this too thoroughly yet, but I will need to very soon.
We have limited resources, so will be looking at a way to automate
anything and everything possible. I'm open to suggestions.
In my initial test instance,
On Mon, Apr 28, 2008 at 09:30:18AM -0400, Steve Totaro wrote:
> On Mon, Apr 28, 2008 at 9:20 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> > On Mon, Apr 28, 2008 at 03:07:46PM +0200, Philipp Kempgen wrote:
> >
> > > On the other hand, nothing's _really_ wrong with rc.local, apart
> > > from what
Steve,
You are correct. I've asked this question before. Now that my sysadmin
group has determined the server they would prefer to support, I wanted to
double check.
We won't be relying heavily on the T1 card. I will also be setting up
H.323 trunking to our Avaya Definity G3R. Although ther
>
> I can dial out to other numbers without issue.
>
> Calling the number from a separate PSTN phone works fine.
I have had such problem last week. from PRI interface (i get busy tone as
fast as i finish typing last digit & hit dial) from analog interface I can
make the call without problem.
On Mon, Apr 28, 2008 at 10:08 AM, Alastair Battrick <[EMAIL PROTECTED]> wrote:
> Steve Totaro wrote:
> > On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick <[EMAIL PROTECTED]>
> wrote:
> >> I have a problem calling a certain number from our PRI line. Calling the
> >> number from a separate PST
> > need auto answer or how else would the audio come out of
> the speakers?
One of the lowest cost solutions I've used for this is a Grandstream BT-200.
You can buy the phone for around $50, it supports auto answer, and it has a
miniature audio jack already on it. Just run the output to an ampli
On Mon, Apr 28, 2008 at 10:16 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
>
> On Mon, Apr 28, 2008 at 9:21 AM, Eve-Ellen Cole <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > I am thinking of going with a Dell PowerEdge 1950 ||| for a new
> > CentOS/Asterisk set up. It will have dual 2.33GHz p
On Mon, Apr 28, 2008 at 9:21 AM, Eve-Ellen Cole <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I am thinking of going with a Dell PowerEdge 1950 ||| for a new
> CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory,
> two 500GB hard drives (presumably mirrored). I also plan to get a Dig
Steve Totaro wrote:
> On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick <[EMAIL PROTECTED]> wrote:
>> I have a problem calling a certain number from our PRI line. Calling the
>> number from a separate PSTN phone works fine.
>>
>> The remote number seems to have some funny call redivert setup, wh
On Mon, Apr 28, 2008 at 9:32 AM, Arthur <[EMAIL PROTECTED]> wrote:
>
> > Make sure you get a "helpful tech" on the phone. Many times they will
> > just dismiss you with "we cannot do that" even though they may be able
> > to.
>
> i always say if you pay your bills you should get the support you di
i suggest a look at digium's hardware compatibility list on thier website.
i would also worry about concurrent calls & thus concurrent recordings, 48
with your actual card which i guess is acceptable load to your hardware but
i am not an expert in this.
recording to disk (even scsi ones) will make
On Mon, Apr 28, 2008 at 9:29 AM, Alastair Battrick <[EMAIL PROTECTED]> wrote:
> I have a problem calling a certain number from our PRI line. Calling the
> number from a separate PSTN phone works fine.
>
> The remote number seems to have some funny call redivert setup, when you
> call it, it answ
The 2 Port card may not provide the number of channels you may need to
do this. I would bump it up to a four port.
I would also look at more HD space. You are fine on RAM memory, if you
need to for budget constraints I would be OK with dropping the RAM and
upping the Hard Drive Space. 2-4 GB of
>
> Make sure you get a "helpful tech" on the phone. Many times they will
> just dismiss you with "we cannot do that" even though they may be able
> to.
i always say if you pay your bills you should get the support you diserve. &
every provider is almost always willing to help out his clients if
On Mon, Apr 28, 2008 at 9:20 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Mon, Apr 28, 2008 at 03:07:46PM +0200, Philipp Kempgen wrote:
>
> > On the other hand, nothing's _really_ wrong with rc.local, apart
> > from what Tzafrir explained.
>
> It can be bad if it actually hides the real pro
-- Executing [EMAIL PROTECTED]:23] NoOp("IAX2/255-4", "Using
CallerID "Alastair Battrick" <255>") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("IAX2/255-4",
"_NODEST=") in new stack
-- Executing [EMAIL PROTECTED]:3] Macro("I
remove callprogress=yes and busydetect=yes
lotusscript wrote:
> Been using the Snom 360 and 190 for a while and decided to try the Cisco
> 7960. The problem I'm seeing is the call terminates between 2:34 and
> 3:00 minutes. This only happens when using Zap channels. Internal
> calls work fine.
Philipp Kempgen schrieb:
> Steve Totaro schrieb:
>
>> I cannot comment on the init scripts because I don't use them.
>>
>> Since nobody chimed in that using rc.local was not optimal, I suspect
>> it is perfectly acceptable and has worked for me just fine on MANY
>> installs.
>
> Tzafrir said rc.
I am thinking of going with a Dell PowerEdge 1950 ||| for a new
CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB
memory, two 500GB hard drives (presumably mirrored). I also plan to get a
Digium TE220B to go with it. (a non-dell server is not an option, but I
am wondering if the
On Mon, Apr 28, 2008 at 9:07 AM, Philipp Kempgen
<[EMAIL PROTECTED]> wrote:
> Steve Totaro schrieb:
>
>
> > I cannot comment on the init scripts because I don't use them.
> >
> > Since nobody chimed in that using rc.local was not optimal, I suspect
> > it is perfectly acceptable and has worked
On Mon, Apr 28, 2008 at 03:07:46PM +0200, Philipp Kempgen wrote:
> On the other hand, nothing's _really_ wrong with rc.local, apart
> from what Tzafrir explained.
It can be bad if it actually hides the real problem.
I asked the OP twice for some additional information, and got no reply
so far.
Again, a reply to my reply. Note to self: stop hitting send before
completing thoughts.
Maybe if you ask the telco to turn off the SLA blocking. It may not
solve the underlying issue but it may allow you to continue inbound
and outbound without service interruption providing it does not drop
an
This may be more helpful as far as Asterisk implementation. Sorry I
cannot be of more help, I have never dealt with this tech.
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
Thanks,
Steve Totaro
On Mon, Apr 28, 2008 at 9:06 AM, Arthur <[EMAIL PROTECTED]> wrote:
> http://www.soft-switch.org/
Bilal,
Sorry to reply to my reply but if you can register multiple accounts
and setup auto answer on one of those accounts, it could work. The
problem is, I am not sure if there are any ATAs that have this
ability.
Thanks,
Steve Totaro
On Mon, Apr 28, 2008 at 9:06 AM, Steve Totaro
<[EMAIL PROTE
Bilal,
No, I do not think that you can make this work. You would obviously
need auto answer or how else would the audio come out of the speakers?
I thought you were talking about overhead paging.
Thanks,
Steve Totaro
On Mon, Apr 28, 2008 at 8:49 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> D
Steve Totaro schrieb:
> I cannot comment on the init scripts because I don't use them.
>
> Since nobody chimed in that using rc.local was not optimal, I suspect
> it is perfectly acceptable and has worked for me just fine on MANY
> installs.
Tzafrir said rc.local wasn't optimal and that's right.
http://www.soft-switch.org/unicall/mfcr2/ch02.html
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Dear Steve,
We have installed Asterisk with Digium card TE110P , install MFC R2 connect to
PSTN (indonesia) using DIG13 MFCR2 siemens EWSD, Germany.
asterisk working normaly, outgoing call ok, incoming call ok. but in central
office /PSTN having SLA(service level alarm). If It happend, all chann
On Mon, Apr 28, 2008 at 01:17:33PM +0100, lotusscript wrote:
>
> A good while back when installing 1.2 there were major issues with UK
> callerid. Asterisk 1.2 didn't recognise the callerid correctly because
> of the way BT sent the information. Sometimes before the first ring or
> just after.
On Mon, Apr 28, 2008 at 3:18 AM, Benjamin Jacob <[EMAIL PROTECTED]> wrote:
>
>
>
> >
> http://www.openvox.com.cn/products_detail.php?genre_id=9&id=28
> >
> > If you can get the bare card, you can use it for
> > timing with a little
> > magic that can be found via google. If not, get one
> >
A good while back when installing 1.2 there were major issues with UK
callerid. Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information. Sometimes before the first ring or
just after. After applying a third party patch we got it to work. We
were afraid t
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No probs with the Snoms. No errors show on the * box
when the line d
Bilal,
I cannot comment on the init scripts because I don't use them.
Since nobody chimed in that using rc.local was not optimal, I suspect
it is perfectly acceptable and has worked for me just fine on MANY
installs. Finally, instead of having to do some fishing in
/var/log/asterisk and /var/log
On Mon, Apr 28, 2008 at 7:40 AM, Arthur <[EMAIL PROTECTED]> wrote:
>
> > Please contact me off-list if you would like to test out my beta
>
> i'd be glad to do that.
>
>
> > It is connected to a TDM PRI. Where are you calling? I pay a penny a
> > minute but would be glad to eat that cost for re
On Mon, Apr 28, 2008 at 7:32 AM, Arthur <[EMAIL PROTECTED]> wrote:
>
> > exten => wardial,1,NoOp(wardialing:
> i didn't mean i wanted a wardialer ! i meant a simple predictive dialer
> (that is no rich features, only to be able to make transfers internaly &
> externaly & to show data of called lead
>
> Please contact me off-list if you would like to test out my beta
i'd be glad to do that.
It is connected to a TDM PRI. Where are you calling? I pay a penny a
> minute but would be glad to eat that cost for real world testing.
I also use PRI links only ... & the thought of testing a wa
>
> exten => wardial,1,NoOp(wardialing:
i didn't mean i wanted a wardialer ! i meant a simple predictive dialer
(that is no rich features, only to be able to make transfers internaly &
externaly & to show data of called lead to agent)
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On Mon, Apr 28, 2008 at 6:37 AM, Arthur <[EMAIL PROTECTED]> wrote:
>
> > can tell you that it is an actual asterisk application (with a conf
> > file) and html for the web interface (and obviously a database).
>
> I'd like to start something similar with python/Mysql under the hood & a web
> page a
On Mon, Apr 28, 2008 at 10:37:29AM +, Arthur wrote:
> >
> > can tell you that it is an actual asterisk application (with a conf
> > file) and html for the web interface (and obviously a database).
>
>
> I'd like to start something similar with python/Mysql under the hood & a web
> page at the
Hi All;
Just would like to know if Boger, UPAM, and Onkyo can
be connected to the FXS ports and take extension from
asterisk?
And why Onkyo is needed if I need to use horn
speakers?
Regards
Bilal
---
> > > bilal ghayyad wrote:
> >
> > > > Steve Totaro wrote:
> > > > > On 4/5
>
> can tell you that it is an actual asterisk application (with a conf
> file) and html for the web interface (and obviously a database).
I'd like to start something similar with python/Mysql under the hood & a web
page at the front end (ajax) ... but so far my time does not help. I wonder
if th
Hi Steve;
The problem related to version zaptel 1.4.10.
I just discover that it might be related to init
script because every thing running on the same machine
if I used svn to get the source and compile it, but
ofcourse I will not get the zaptel 1.4.10 version
using svn (maybe it depends on the
Hi,
Have you looked at RTCP statistics ?
As fas as I know, many IP Phones provides such data and Asterisk can store
some of these data in CDRs.
This won't help to predict which quality for future calls but if it can
somehow help.
Cheers
___
-- Bandwidt
>
http://www.openvox.com.cn/products_detail.php?genre_id=9&id=28
>
> If you can get the bare card, you can use it for
> timing with a little
> magic that can be found via google. If not, get one
> with an FXO or
> FXS and you will add a little flexibility and have
> real hardware
> timing.
>
>
--- bee-beeep <[EMAIL PROTECTED]> wrote:
> It works fine in every case, with disabling transfer
> in Dial() options
>
> 2008/4/25 Grey Man <[EMAIL PROTECTED]>:
>
> > > > > Thanks to your answers, but i found more
> beautiful way to do this -
> > > > > there is some system variable
> __TRANSFER_
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