Olle E. Johansson wrote:
snip
Everyone points to capi and, back to the start of my reply, it seems
expensive for personal
use...
A passive AVM Fritz card is somewhere around € 100
/t
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I'm connecting and can place calls to and from my SIP phone that is
behind a firewall, can hear audio from the SIP on the PSTN line but
can't hear audio on the SIP phone from the PSTN line. Anyone else
experience this?
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[EMA
Noticed that I can adjust the number if frames/packet on the GrandStream
phone. Can * do the same?
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Not sure if it's alright to talk about this here???
compiled the STUN server from Vovida on RedHat 7.3. Looks simple to
configure. It isn't starting...it tries to for a long time and then just
craps out. Here is my config:/etc/sysconfig/stund
#!/bin/echo Not to execute.
# Path to stund
STUND=/usr
On Wed, 3 Sep 2003, Jay Tyndall wrote:
> I am receiving calls via a Netjet-S card on asterisk, and I notice that
> whenever I am talkimng to someone, if their voice is loud enough, sometimes
> asterisk generates a DTMF Tone as they speak. that is played to me. (Caller
> doesn't hear it).
>
> A
On Wed, 3 Sep 2003, Jay Tyndall wrote:
> I am using a Netjet-s ISDN Card, and am having some trouble dialling out
> (Incoming Works Fine).
...
> I get the following when diallingout:
> -- Starting simple switch on 'Zap/2-1'
>-- Executing Dial("Zap/2-1", "Modem/ttyI0/04||Ttm") in new
Hi all,
Currently trying to get asterisk to dial out with an Internet Line Jack card,
however, it does not use the pots line, only on the line it dials out of. This
is similar to the previous thread/posting "Asterisk won't answer pstn ring",
but I didn't find any follow up to get it working.
My a
Hello all,
I have been asked to help in the design of a lab for a
telecommunications technology course (third year students) to teach VoIP
technologies. Here is a cut n' paste of an email I received from one of
my instructors
"A very important consideration is the numbering plan and if telephone
On Tue, 2003-09-02 at 22:50, [EMAIL PROTECTED] wrote:
> On Tue, Sep 02, 2003 at 09:35:34PM -0600, Gavin Hollinger wrote:
> > >> > Sounds like an IRQ issue.
> > > Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but
> >
> > Looks ok?
> >
> >
> > cat /proc/interrupts
> >
It only core's when u use the gatekeeper component, due to the way pwlib
deals with memory allocation. This is going to take quite a lot of
trying various different incantations to fix, unfortunately I cannot
justify dedicating that kind time, at this point.
Sorry,
Jeremy McNamara
Martin P
On Tue, Sep 02, 2003 at 09:35:34PM -0600, Gavin Hollinger wrote:
> >> > Sounds like an IRQ issue.
> > Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but
>
> Looks ok?
>
>
> cat /proc/interrupts
>CPU0 CPU1
> 0: 9228581476395IO-APIC-edge timer
>> > Sounds like an IRQ issue.
> Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but
Looks ok?
cat /proc/interrupts
CPU0 CPU1
0: 9228581476395IO-APIC-edge timer
1: 0 4IO-APIC-edge keyboard
2: 0 0
On Tue, Sep 02, 2003 at 10:10:17PM -0500, Peter Pauly wrote:
> On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
> > > correctly from X-lite but nothing else happens - no audio is
> > > heard. My understanding is that I should hear some sort of
> >
> > I am using x-lite with the ast
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote:
> > correctly from X-lite but nothing else happens - no audio is
> > heard. My understanding is that I should hear some sort of
>
> I am using x-lite with the asterisk demo no problem. All I modified was
> sip.conf
>
> Is the aster
That's OK... we all start somewhere. And frankly, I'm not that far behind
you. It is interesting to see so many people who are new to Asterisk join
the list: I think it suggests a good future for Asterisk.
Tim
On Tuesday 02 September 2003 08:36 pm, Frank Latini wrote:
> Thanks..that was the q
Thanks..that was the question...too new to understand.
- Original Message -
From: "Timothy Soos" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 02, 2003 22:27
Subject: Re: [Asterisk-Users] Sip Software from Nero Folk?
> On Tuesday 02 September 2003 08:19 pm, Frank L
Hi again, after reading more messages in the list. I get it! Thanks for
the non-flames, etc.
Frank
- Original Message -
From: "Frank Latini" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 02, 2003 22:19
Subject: Re: [Asterisk-Users] Sip Software from Nero Folk?
>
* = Asterisk symbol = Asterisk = reference to Asterisk PBX
-Original Message-
From: Frank Latini [mailto:[EMAIL PROTECTED]
Sent: Tue 9/2/2003 10:19 PM
To: [EMAIL PROTECTED]
Cc:
Subject: Re: [Asterisk-Users] Sip Software from Nero Folk?
On Tuesday 02 September 2003 08:19 pm, Frank Latini wrote:
> Hi all,
>
> New to the list. We are going to begin testing various voip gateways. I
> am trying to understand the reference to * in this thread. Is there a rule
> of the list that I need to be aware of ? Do not want to breech the
> e
Hi all,
New to the list. We are going to begin testing various voip gateways. I am
trying to understand the reference to * in this thread. Is there a rule of
the list that I need to be aware of ? Do not want to breech the etiquette
of the list.
Thanks
Frank
- Original Message -
Fro
On Tue, Sep 02, 2003 at 03:37:03PM -0600, Gavin Hollinger wrote:
> > Sounds like an IRQ issue.
>
> Perhaps. If that was the case, I should see an error somewhere right?
>
> Where could I look?
>
> Gavin
Sometimes I find /proc/interrupts useful. if you see a lot of irqs (but not
for sound devic
Hi,
I am receiving calls via a Netjet-S card on asterisk, and I notice that
whenever I am talkimng to someone, if their voice is loud enough, sometimes
asterisk generates a DTMF Tone as they speak. that is played to me. (Caller
doesn't hear it).
Any ideas how to stop this?
___
Using Digium's "Asterisk Developer's Kit (TDM)",
I've been trying to make an outside call by copying
sample.call to /var/spool/asterisk/outgoing.
I want the VoiceMailMain to run when the call is
answered.
The call is made correctly but, as
you probably know, the application starts as soo
Hi,
I am using a Netjet-s ISDN Card, and am having some trouble dialling out
(Incoming Works Fine).
TRUNK=Modem/ttyI0
exten => _90X,1,Dial(${TRUNK}/${EXTEN:1}||Ttm)
exten => _90X,2,Congestion
I get the following when diallingout:
-- Starting simple switch on 'Zap/2-1'
-- Exe
On Tue, 2003-09-02 at 09:24, Martin Pycko wrote:
> This happens only on relaod. You can disable reload routine in chan_h323.c
> ...
Thanks. I'll give it a try.
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On Monday 01 September 2003 02:18 pm, Tarun Banka wrote:
> Hello All,
>
> I would like to know the most commonly used IP Phones with
> Asterisk PBX. Your experience will help me in taking a right
> decision to buy IP phones.
>
> Does anyone has experience with Telstrat i2732 IP Telephone and
> SipP
On Tue, 2003-09-02 at 17:11, Gavin Hollinger wrote:
> > Note that this adds anything much to your problem. But I wanted to note
> > that not all systems have hard times with systems working. This system
>
> Thanks for taking the time to do that for me. We are building a bigger
> system that will
unsubscribe
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> Note that this adds anything much to your problem. But I wanted to note
> that not all systems have hard times with systems working. This system
Thanks for taking the time to do that for me. We are building a bigger
system that will be under heavier load with lots of 56k data. However,
your in
Kd,
DID is handled like an extension:
exten => 901212,1,Dial(SIP/[EMAIL PROTECTED])
A quick google search of the list was all I needed to find this...
"site:lists.digium.com DID routing"
-wade
-Original Message-
From: Kekin Dand [mailto:[E
i run it on freebsd. it has worked flawlessly. i prefer
pf config file syntax over any of the others: ipfw[12], ipfilter
and the linux variants. the pflog+pftcpdump feature is handy
for seeing what your filter is denying. the only thing i miss
in using it is dummynet.
On 2003.09.02 17:17:1
On Tue, 2003-09-02 at 16:17, Gavin Hollinger wrote:
> TE410P - intermittent one way audio forcing reboot multiple times per day.
>
> > but have thought of restarting asterisk at cron time. If someone's on
> > the phone at that time they're only talking to themselves anyway.
>
>
> Yeah, that woul
On Tue, 2 Sep 2003, Jon Pounder wrote:
> At 05:17 PM 9/2/2003 -0400, you wrote:
> >Hello.
> >Trying firewalls out.
> >Anyone had any success with an Openbsd PF firewall ?
>
> works for us, seems fairly simple to configure, and tamper resistant since
> it can run in bridge mode with no externally
> Sounds like an IRQ issue.
Perhaps. If that was the case, I should see an error somewhere right?
Where could I look?
Gavin
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Hello Everyone,
I am new to asterisk and linux too, I managed to installed
asterisk on redhat8 with the help of mailing list archives and Handbook guide.
I configured 2 SIP phones (grandstream) and it is working
fine internally.
We have T1 Line coming in with block of 200 DID Num
> correctly from X-lite but nothing else happens - no audio is
> heard. My understanding is that I should hear some sort of
I am using x-lite with the asterisk demo no problem. All I modified was
sip.conf
Is the asterisk server and your x-lite client on the same LAN segment?
Is all iptables and
Sounds like an IRQ issue.
bkw
On Tue, 2 Sep 2003, Gavin Hollinger wrote:
> TE410P - intermittent one way audio forcing reboot multiple times per day.
>
> > but have thought of restarting asterisk at cron time. If someone's on
> > the phone at that time they're only talking to themselves anyway.
At 05:17 PM 9/2/2003 -0400, you wrote:
Hello.
Trying firewalls out.
Anyone had any success with an Openbsd PF firewall ?
works for us, seems fairly simple to configure, and tamper resistant since
it can run in bridge mode with no externally visible ips, so it is
impossible for an attacker to ga
try system,"/usr 01 on"
or system("/usr .. on")
Martin
On Tue, 2 Sep 2003, Josh Edwards wrote:
> Question below, here is the file in question
> exten => 9,1,system,/usr/local/bin/hetest 01 on
> exten => 9,2,system,/usr/local/bin/hetest 02 on
> exten => 9,3,system,/usr/local/bin/hetest 03 on
TE410P - intermittent one way audio forcing reboot multiple times per day.
> but have thought of restarting asterisk at cron time. If someone's on
> the phone at that time they're only talking to themselves anyway.
Yeah, that would work, except I am also having intermittent problems with
the sam
Hello.
Trying firewalls out.
Anyone had any success with an Openbsd PF firewall ?
Regards...Martin
--
Good news. Ten weeks from Friday will be a pretty good day.
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On Tue, 2003-09-02 at 15:48, Josh Edwards wrote:
> Nothing, it is a shell script that runs another program. do I need to
> have it return something?
Yes, all applications should return a result code after finishing.
Usually 0 means success or no errors and anything else is some form of
error indic
Nothing, it is a shell script that runs another program. do I need to have it return something?
Josh
>From: Steven Critchfield <[EMAIL PROTECTED]>
>Reply-To: [EMAIL PROTECTED]
>To: [EMAIL PROTECTED]
>Subject: Re: [Asterisk-Users] extensions.conf issue
>Date: Tue, 02 Sep 2003 15:11:34 -0500
>
I have been using X-Lite on FWD without any troubles
and recently became interested in trying asterisk.
I am able to register from X-Lite and dial a number -
I've tried dialing some of the sample numbers in the sample
extentions.conf file, like 500 and 1234, they appear to dial
correctly from X
Has anyone had any success using a Cisco 7905G phone with Asterisk?
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On Tue, 2003-09-02 at 14:54, Josh Edwards wrote:
> Question below, here is the file in question
> exten => 9,1,system,/usr/local/bin/hetest 01 on
> exten => 9,2,system,/usr/local/bin/hetest 02 on
> exten => 9,3,system,/usr/local/bin/hetest 03 on
> exten => 9,4,system,/usr/local/bin/hetest 04 on
> e
Try System("/usr/local/bin/hetest 01 on")
bkw
On Tue, 2 Sep 2003, Josh Edwards wrote:
> Question below, here is the file in question
> exten => 9,1,system,/usr/local/bin/hetest 01 on
> exten => 9,2,system,/usr/local/bin/hetest 02 on
> exten => 9,3,system,/usr/local/bin/hetest 03 on
> exten => 9,
Question below, here is the file in question
exten => 9,1,system,/usr/local/bin/hetest 01 onexten => 9,2,system,/usr/local/bin/hetest 02 onexten => 9,3,system,/usr/local/bin/hetest 03 onexten => 9,4,system,/usr/local/bin/hetest 04 onexten => 9,5,system,/usr/local/bin/hetest 05 onexten => 9,6,system
Hi Olle,
the cheapes CAPI card is the passive AVM Fritz Card PCI. It's nice
to start playing, but for a production system (isdn pbx) i can only
recommend the active Eicon Diva Server cards (which support EC).
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13
Title: Message
Has anyone been able
to exit voicemail2 to get to voicemailmain2 in order to check voicemail messages
from a remote phone? I can dial a "0" while my voicemail intro plays, and *
dumps the call and says it was sent into invalid extension "o". I set up
extension "0", and I can c
Tomas Prybil wrote:
max power wrote:
Spent that last week or so trying to get isdn4linux working. how
do I link ttyIO to asterisk?I cannot dial out or dialin. I can
see the call coming in in /var/log/messages.
Has anyone any tips? I am not familar with isdn4linux.
What kind of IDDN
max power wrote:
Spent that last week or so trying to get isdn4linux working. how do I link ttyIO to asterisk?I cannot dial out or dialin. I can see the call coming in in /var/log/messages.
Has anyone any tips? I am not familar with isdn4linux.
Hi.
What kind of IDDN device are
Try canreinvite=no
Martin
On Tue, 2 Sep 2003, Zac Sprackett wrote:
> I'm having a problem getting outbound trunking to work using asterisk
> and an external SIP FXO.
>
> 7 digit dialing produces the following output:
>
> -- Executing Dial("SIP/mitel-fe17", "SIP/[EMAIL PROTECTED]") in new stack
Well put.
On Tue, 2 Sep 2003, John Todd wrote:
> At 11:42 AM -0500 9/2/03, Brian West wrote:
> >
> >http://bugs.digium.com/bug_view_page.php?bug_id=149
> >
> >bkw
> >
> >On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
> >
> >> On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
> >> Brian West <[EMAIL PRO
At 11:42 AM -0500 9/2/03, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=149
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
Brian West <[EMAIL PROTECTED]> wrote:
> I opened a request on bugs.digium.com for this feature. The 6k
I'm having a problem getting outbound trunking to work using asterisk
and an external SIP FXO.
7 digit dialing produces the following output:
-- Executing Dial("SIP/mitel-fe17", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/mediatrix-1204-645e answered SIP/mitel-fe17
On Tue, 2003-09-02 at 12:07, John Todd wrote:
> >On Tue, 2003-09-02 at 02:27, John Todd wrote:
> >> Phil -
> >> Here are my "generic" notes and reminders for Asterisk on Debian.
> >> These may be hacks; your mileage may vary.
> >>
> >> debian asterisk install notes:
> >> - in asterisk/Makef
On Tue, 2 Sep 2003, John Todd wrote:
>I'm trying to make the MWI indicators on my client's Vodavi Starplus
>DHS phones work. The actual signalling - in-band DTMF from the ZAP
>interfaces directly to the PBX system - works fine. I can manually
>tell asterisk to send "#9610" as DTMF and voila, t
Spent that last week or so trying to get isdn4linux working. how do I link ttyIO
to asterisk?I cannot dial out or dialin. I can see the call coming in in
/var/log/messages.
Has anyone any tips? I am not familar with isdn4linux.
Max...
__
On Tue, 2003-09-02 at 02:27, John Todd wrote:
Phil -
Here are my "generic" notes and reminders for Asterisk on Debian.
These may be hacks; your mileage may vary.
debian asterisk install notes:
- in asterisk/Makefile: added "-I/usr/local/ssl/include" to CFLAGS line
- in asterisk/res/Makefil
My point as that you would need to pass the fax traffic through via the f
extension to nother port which would be back to back connected to something
that is a faxmodem and talks to hylafax
Another option are the Eicon DIVA Server cards which could do the needed stuff
via CAPI.
On Tuesday 02 Se
http://bugs.digium.com/bug_view_page.php?bug_id=149
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
> On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
> Brian West <[EMAIL PROTECTED]> wrote:
>
> > I opened a request on bugs.digium.com for this feature. The 6k and 8k
> > codecs are very impressive a
HylaFAX needs to connect to a modem, and the modem in turn needs to
connect to a phoneline. This phoneline has to be "real" so you can't
use a dummy driver. The TDM400P or T100P/E100P/T400P/TE410P+channel
bank would be the only suitable (and supported) choices for analog modem
connections.
-wade
Well this debug desn't show the bad call setup. And furthermore all
commands are accepted by the asterisk/UA.
Martin
On Mon, 1 Sep 2003, Andrew Joakimsen wrote:
> There might be some other stuff mixed in there as well, 64.36.104.205 is
> asterisk and 64.36.104.206 is the DTA
>
> 11 headers, 2 li
I am testing out vmail.cgi
I can listen to my messages, but I can't forward them
to another user.
I get the following error message:
Software error:
Invalid new mailbox
That doesn't tell me much, so I'm hoping that somebody
will be able to help me out.
Thank you for your time.
__
Do the ports have to be real (like XP100) or can you use the ztdummy (the
dummy zapatel driver) for the ports required for hylafax?
Lee Goodman
- Original Message -
From: "Michael Bielicki" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 02, 2003 4:18 AM
Subject: Re: [
This happens only on relaod. You can disable reload routine in chan_h323.c
...
Martin
On 1 Sep 2003, Michael wrote:
> I'm running the CVS from last week and from day one (over 4 months now)
> I've had this problem where asterisk core dumps when using chan_h323.
>
> It appears to be a problem wit
On Tue, 2 Sep 2003, John Todd wrote:
> >I'm trying to make the MWI indicators on my client's Vodavi Starplus
> >DHS phones work. The actual signalling - in-band DTMF from the ZAP
> >interfaces directly to the PBX system - works fine. I can manually
> >tell asterisk to send "#9610" as DTMF and
On Tue, 2003-09-02 at 10:00, [EMAIL PROTECTED] wrote:
> php is not just a web scripting language anymore. it has been used in
> other ways for quite a while now. it works nicely from the command line,
> can be used with ncurses and with gtk. there are several well-known
> respectable large proje
Your configs look ok. All you need is BNC to RJ45 converter (I think the
standard is G.703)
regards
Martin
On Tue, 2 Sep 2003, Langley, Sean wrote:
> Dear Telcotype Braniacs,
>
> I have tried doing a google search to find out what this switch looks like, what the
> physical interface is, but ha
At 12:29 PM 9/2/2003 +0100, you wrote:
I'm new
in * and I would like to know what version of the Linux kernel or RedHat
Distribution do you recomend.
Redhat 9 works perfectly. Install with the kernel sources and devel
libraries, and the developers software, i.e. gcc, and upgrade to most
recent rp
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT)
Brian West <[EMAIL PROTECTED]> wrote:
> I opened a request on bugs.digium.com for this feature. The 6k and 8k
> codecs are very impressive also.
>
> bkw
>
Where can I see the status of this request?
[]'s
Eduardo
_
Dear Telcotype Braniacs,
I have tried doing a google search to find out what this switch looks like, what the
physical interface is, but havn't been successful. I am quite new to the ISDN world
so I'm not sure what to expect when I see this switch. Does anyone have any
experience connecting t
I opened a request on bugs.digium.com for this feature. The 6k and 8k
codecs are very impressive also.
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
> Hello,
>
> I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps.
> With asterisk, what's the bit rate used
php is not just a web scripting language anymore. it has been used in
other ways for quite a while now. it works nicely from the command line,
can be used with ncurses and with gtk. there are several well-known
respectable large projects out there built upon php. i usually find that
php's bigge
On Tue, 2003-09-02 at 02:27, John Todd wrote:
> Phil -
>Here are my "generic" notes and reminders for Asterisk on Debian.
> These may be hacks; your mileage may vary.
>
> debian asterisk install notes:
> - in asterisk/Makefile: added "-I/usr/local/ssl/include" to CFLAGS line
> - in asterisk/r
I noticed in the documentation that it is possible to use isdn4linux compatible
hardware with *.
I have a Dynalink 6692 PCI card which is ISDN4Linux compatible. How do I use it
within * ?
I would be grateful for any help.
Thanks
Tim
___
Asterisk-
But, you could use a third-party fax thingamajig and I'm sure connect it
to * for a good UM solution.
Just pass it to hylafax and you fly, but it requires some planning cause you
will need ddouble the amount of ports plus the fax devices for hylafax
Interesting - please, do you have time to elabo
Redhat 8 and 9 have both worked fine for me..
I have an install guide at http://members.lycos.co.uk/wipe_out/asterisk
If you are intersted..
Later- Original Message -From: "Francisco Mesquita" <[EMAIL PROTECTED]>Date: Tue, 2 Sep 2003 12:29:23 +0100To: <[EMAIL PROTECTED]>Subject: [Asterisk-U
Hello,
I've read about speex's bit rate. Speex can work from 2.15kbps to 44.2kbps.
With asterisk, what's the bit rate used by speex? Is it possible to have
asterisk using speex at less than 10kbps of bit rate? If not, Is it dificult to
implement?
thanks in advance
Eduardo
__
Hi,
I'm new in * and I would like to know what version
of the Linux kernel or RedHat Distribution do you recomend.
Best regards,
Francisco Mesquita
chan_h323 is built into asterisk. Check the /usr/src/asterisk/channels/h323
directory for more info.
- Original Message -
From: "Phillip Britt" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 02, 2003 1:12 PM
Subject: [Asterisk-Users] H.323 Support
Hi,
I am curren
Hi,
I am currently using Asterisk and want to add H.323 support for talking to
our gateway routers, which use gnkgk
Is the package "Asterisk-oh323" the right thing to use, or are there better
ways of achieving h.323 support in Asterisk.
Thanks,
Phil
_
I tried specifying rxgain & txgain,
copied the format some some message on asterisk-users
Result was asterisk bombed out & didn't even load
due to not being able to understand the config file ..
what's the exact syntax that works??
cheers
Dave
- Original Message -
From: "Fredrik Hedberg"
On 01 Sep 2003 23:23:53 -0500
Steven Critchfield <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
On Mon, 2003-09-01 at 22:23, Dave Packham wrote:
http://www.nero.com/us/631911127302064.html
Have you all seen this?
Its a SIP softphone put o
For some reason, chan_h323 ignores the callerid and puts your IP address in
instead. I've modded my chan_h323 to use the caller's id instead. Trival
change but I'm guessing there's a reason why it isn't so in the first place.
Anyone know why?
Adam Hart
- Original Message -
From: Rattana
I've been having the same problem too, except for me it only happens
occasionnally.
I'm not 100% sure of this, but it seems that for very local calls (eg across
the city) I never get echo. For calls that go longer distance (say 500km or
more), or through some closer call centres, I'm getting the
I've been having the same problem too, except for me it only happens
occasionnally.
I'm not 100% sure of this, but it seems that for very local calls (eg across
the city) I never get echo. For calls that go longer distance (say 500km or
more), or through some closer call centres, I'm getting the
Hi,
Ok, the phones are working and seem to be loading the correct info from the
tftp server. However, I am unable to make them perform any functions
(calling another extension, going to voicemail, etc.). I do not have any
telephony interface installed yet, only a single ethernet card. Do I need
I downlaoded it and tried it, SIPPS. Nice featureful sip client, however, I
haven't been able to get it to pass dtmf to *. I don't know if this is a
software restriction or not, but I have emailed nero asking them for their
opinion of this, as it is, in my case, a LARGE restriction when trying t
On Mon, 2003-09-01 at 22:23, Dave Packham wrote:
> http://www.nero.com/us/631911127302064.html
>
>
> Have you all seen this?
>
> Its a SIP softphone put out by the people that do the CD burning software Nero...
>
> Check it out it works with *
And the benefit of using a commercial software
For some reason, chan_h323 ignores the callerid and puts your IP address in
instead. I've modded my chan_h323 to use the caller's id instead. Trival
change but I'm guessing there's a reason why it isn't so in the first place.
Anyone know why?
Adam Hart
- Original Message -
From: Rattana
On Tue, 1 Sep 2003, Tarun Banka wrote:
> One quick question. Does anyone has experience implementing
> unified messaging (UM) using Asterisk. Does Asterisk has support
> for UM ?
Not really. Half of it works. If you mean by UM (Voicemail integrated to
your email box, and then send an SMS/Page
http://www.nero.com/us/631911127302064.html
Have you all seen this?
Its a SIP softphone put out by the people that do the CD burning software Nero...
Check it out it works with *
Dave
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On Monday 01 September 2003 03:51, Mickey Binder wrote:
> How do I change the dialplan runtime, if I for example wants all
> calls on the main number to be answered by a voicemail (when it is
> out-of-office hours).
> I want to be able to change the configuration by pressing a DTMF
> combination e.
Brancaleoni Matteo
Thank you very much for your pointers.
I wrote a little PHP function which read an input
from http://stdin
I can extract it and choose a needed value.
Now a variable of PHP-based-AGI script
contents a dialed extension :)
Romsun Pramudito
--
Hi all,
Does anyone have a working IXJ / Dial in config they'd lke to share with me?
Thanks,
Andrew Griffiths
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Actually, I do have that. i've tried inband, as well as rfc2883. Neither
work. I'm going back and forth with ahead software on the issue, and
they're doing a little bit of looking into it.
Doesn't even work when clicking on the numbers, as required by the software,
as someone else pointed out,
-Original Message-
From: Tomas Prybil [mailto:[EMAIL PROTECTED]
Sent: 2. september 2003 10:50
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Change include contexts runtime
Mickey Binder wrote:
>>It looks like it. With DBput and DBget im able to change the variable
values
>>and then
Mickey Binder wrote:
It looks like it. With DBput and DBget im able to change the variable values
and then branch to different contexts with GotoIf. Now I just need to
implement the right logic for the different situations.
And maybe be able to get some sort of feedback to the users.
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