On Sunday 31 August 2003 16:49, Michiel Betel wrote:
> The db entries persist on reload, on a restart (or crash...) they are
> gone...
Are you perhaps running Asterisk as a user other than root? Sounds like
you might not have permission to write to /var/lib/asterisk/.
-Tilghman
Yes, keep up the good work!
On Sunday, August 31, 2003, at 09:24 AM, Brian West wrote:
I have added support for enum looks for iax,iax2 and h323. So far in
my
testing it has worked perfect. (note: you need to strip the + for iax
and
iax2 calls or they will fail. h323 will accpet the + but I
OOOPS Indeed! My fault... They do persist if you the system it
correctly.
Sorry, Michiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: maandag 1 september 2003 5:12
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DBSa
Hi there
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.g. *82. Can't figure out whether it is necessary to chan
Hi,
I use with asterisk gnugk a gatekeeper for h323
client.
I don't understand why asterisk can't have the
H323-ID (callerID).
In the gatekeeper's monitor I have this H323-ID but
not in asterisk.
Does anyone know something about it, or how can I
send a caller ID to asterisk ?
Ra
Hi all,
i have a few questions about PRI/ISDN:
1. Are "supplementary services" like conferencing, call brokering or call
forwarding supported by * ?
2. Is there a way to switch calls "transparent" through * from one port to
another port ?
3. Is it possible to configure the
I have the same problem
- Original Message -
From: "Mickey Binder" <[EMAIL PROTECTED]>
To: "Asterisk maillist (E-mail)" <[EMAIL PROTECTED]>
Sent: Monday, September 01, 2003 10:51 AM
Subject: [Asterisk-Users] Change include contexts runtime
> Hi there
>
> How do I change the dialplan runt
Hi all,
this message occurs if i was connected or not:
WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno
102 (Response)
If i was connected, the call will be disconnected after a few seconds.
What does it means ? I don't see an
Hey,
Has anyone managed to get X-Lite and Asterisk to play nicely using iLBC yet?
Later..
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Asterisk-Users
I am in Australia and I have a friend in the
USA. We both have asterisks installed and dial tone.
What is the easiest way to set us both up so we
each have an extension number to call each other on?
Also, There is a lot of information in the
Asterisk's manual. Has anyone produced a qui
Best bet is to setup an IAX trunk between the two * servers..
I have an install guide but if you are setup already than you are already past that
stage.. :)
its at http://members.lycos.co.uk/wipe_out/asterisk
> I am in Australia and I have a friend in the USA. We both have asterisks installe
> 1. Are "supplementary services" like conferencing, call brokering or call
> forwarding supported by * ?
Conferencing (check MeetMe application), cal brokering ??? call forwarding
you can do that by having a little script in extensions.conf (unless
you're using FXS ports, where you can use *
Hi Martin,
in EuroISDN it is possible to hold the call and take another. This is part
of ISDN suplementary services.
Or make three-way-calling ...
And the question was if the * if you dial in
like your config:
exten => _X.,1,Dial,Zap/g2/${EXTEN}
these services will be tranfered to the Network E
Hi,
I am new to Asterisk and wanted to ask a question concerning PSTN trunking. Is there a way to have DID's sent over IP to a switch? I know if One switch has traditional PSTN like a PRI this can be done, but is there a service provider offering this so I dont have to buy any tradtional PSTN
You can use timeofday associations on "include =>" for the most elegant
solution or, you can also use multiple extensions.conf and "#include" (not
the same as "include =>" ) and asterisk -rx reload.
Mark
On Mon, 1 Sep 2003, Rattana BIV wrote:
> I have the same problem
>
> - Original Message
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
>
On Mon, 1 Sep 2003, jim b wrote:
> I am new to Asterisk and wanted to ask a question concerning PSTN
> trunking. Is there a way to have DID's sent over IP to a switch? I know
> if One switch has traditional PSTN like a PRI this can be done, but is
> there a service provider offering this so I dont
>> read /usr/src/linux/Documentation/networking/netlogging.txt
> It doesn't exist there in 2.4.22-rc4.
After searching the net, I was able to find many references to this
document but no links to the document itself. None of my Linux systems
had it either. I would appreciate if someone could pos
Thomas,
> in EuroISDN it is possible to hold the call and take another. This is part
> of ISDN suplementary services.
> Or make three-way-calling ...
Suplementray services are not supported. But then, you'll find almost
every single feature to be available in asterisk itself.
> And what is with
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 lines
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.36.104.205:5060;branch=z9hG4bK471af161
From: "asterisk" ;tag=as17328ab1
To:
Contact:
Hi Gavin,
sorry, i was assuming that everybody uses SuSE kernels (which have
a lot of useful stuff already built in)... :-)
here is the URL for the netconsole patches:
http://www.kernel.org/pub/linux/people/mingo/netconsole-patches
best regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Jungh
On 1 Sep 2003, Klaus-Peter Junghanns wrote:
> here is the URL for the netconsole patches:
> http://www.kernel.org/pub/linux/people/mingo/netconsole-patches
No work for me, instead:
http://people.redhat.com/mingo/netconsole-patches/
Is that what you meant?
--
Mike Ciholas
the link is broken.
use http://people.redhat.com/mingo/
see ya.
matteo.
Il lun, 2003-09-01 alle 18:40, Klaus-Peter Junghanns ha scritto:
> Hi Gavin,
>
> sorry, i was assuming that everybody uses SuSE kernels (which have
> a lot of useful stuff already built in)... :-)
>
> here is the URL for t
Hi List
I have one question about MGCP in asterisk. I have a media gateway, and I
want to have asterisk to work with the media gateway. As I was told that the
media gateway can communicate with the switch via standard interface
MGCP/ICGP. Question is if the asterisk MGCP supports such MGCP messa
On Mon, 2003-09-01 at 18:40, Klaus-Peter Junghanns wrote:
> Hi Gavin,
>
> sorry, i was assuming that everybody uses SuSE kernels (which have
> a lot of useful stuff already built in)... :-)
>
> here is the URL for the netconsole patches:
> http://www.kernel.org/pub/linux/people/mingo/netconsole-p
It was submited to bugs and added to the base isntall of * today.
bkw
On Mon, 1 Sep 2003, Paul Cheng wrote:
> Yes, keep up the good work!
>
> On Sunday, August 31, 2003, at 09:24 AM, Brian West wrote:
>
> > I have added support for enum looks for iax,iax2 and h323. So far in
> > my
> > testing
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 with pwlib and the console, but I'm not sure
how to read the below output from gdb. I can start Asterisk just fine
and chan_h323
Are you using the recommended pwlib and openh323 versions?
bkw
On Mon, 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 with pwlib and the co
On Mon, 2003-09-01 at 11:19, Brian West wrote:
> Are you using the recommended pwlib and openh323 versions?
>
Yes.
lrwxrwxrwx1 root root 12 Aug 17 20:39 pwlib ->
pwlib-1.4.11
lrwxrwxrwx1 root root 15 Aug 17 20:01 openh323 ->
openh323-1.11.7
Michael
_
Hello,
I am trying the PrePaid IVR system from
http://www.bkw.org/~brian/agi-ccard.agi
and when I am dialing extension and somebody pickup
i got this wornings and then the call ius disconnected.
WARNING[114696]: File chan_sip.c, Line 399
(__sip_xmit): sip_xmit of 0x80ceafc (len 630) to 0
hi ..
i have the exact same problem you have
..
seems to be related to Budgettone phones in my
prob.
I *tried* selling an asterisk exchange to a
client
and today he phoned telling me he is very
unsatisfied
& I risk being thrown out .. suggestions would
be
welcome! i've tried *everything* t
hi everybody
is it posible to configure a RAS with a digium card in a linux box?
thanks
--
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administración servidores y servicios de internet
red de datos
universidad del cauca
http://www.unicauca.edu.co/~santiago/llaves/santiago_pub.asc
hay 10 tipos de personas, las
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
SipPhone IP phones. Are these compatible with the ASterisk ?
Any kind o
Hello,
One quick question. Does anyone has experience implementing
unified messaging (UM) using Asterisk. Does Asterisk has support
for UM ?
Thanks,
Tarun
___
Medicine meets Marketing; Dr. Swati Weds Jayaram.
Rediff Matchmaker strikes another inte
Sipphones are really grandstream budget-tone phones.
we also sell these phones at competative rates. private
offline email if you would like more info..
yes, they work with *. W
john brown
On Mon, Sep 01, 2003 at 08:18:51PM -, Tarun Banka wrote:
> Hello All,
>
> I would like to know the
On Sun, 31 Aug 2003, Josh Edwards wrote:
>
> Are there any examples for ther psql or agi scriptscan I use php
> with
> agi
You most certainly can, but I recommend something more efficient like
c++ or perl, at least for any backend functions.
That said, if you insist on using PHP for th
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>:
Jeremy> What part of "IN OTHER WORDS: Run Open H.323 v1.11.7, nothing
Jeremy> newer, nothing older if u want this to work." don't you
Jeremy> understand?
Well, I was trying to find out (politely) about some things. Please
allow me to paste
- Original Message -
From: "Josh Roberson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 02, 2003 7:22 AM
Subject: Re: [Asterisk-Users] Sip Software from Nero Folk?
> I downlaoded it and tried it, SIPPS. Nice featureful sip client, however,
I
> haven't been able
TE410P - one way audio, after "rpm -qa" on RedHat 9
After much trouble, I can consistently reproduce this error.
Anyone have any idea why all Zap channels on my TE410P would develop one
way audio after "rpm -qa" has been run while asterisk is running?
If I reboot, run "rpm -qa" first then load a
Josh,
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 have played with SIPPS as well. DTMF worked for me, but only when
using the mouse to click on the keys of the virtual keypad :(
Reported it back to Ahead.
I don't
That sounds like a brilliant idea, I will try it right away!
-Original Message-
From: Tilghman Lesher [mailto:[EMAIL PROTECTED]
Sent: 2. september 2003 05:05
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Change include contexts runtime
On Monday 01 September 2003 03:51, Mickey Bind
On Tue, 2003-09-02 at 08:39, Gavin Hollinger wrote:
> TE410P - one way audio, after "rpm -qa" on RedHat 9
>
> After much trouble, I can consistently reproduce this error.
>
> Anyone have any idea why all Zap channels on my TE410P would develop one
> way audio after "rpm -qa" has been run while as
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