is there anyone out there with a running & functional system?
thx
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Great. That is a configuration I have not heard of before, but again
different countries do things differently. Good luck with the ISDN part. I
haven't gotten ISDN and Asterisk working here, yet, primarily because ISDN
is configured differently in the US and not widely used, so if a solution
exi
Hello!
I've configured Music on Hold in asterisk, the only, most certainly, stupid
problem I have is, which DTMFs to send to activate and deactivate it.
If I use the cli, I can establish a call with originate. With the "misdn
send digit" command I can send a number of digits to the other pa
Hello Wilton!
OK, now it works. The ISDN port is for voice telephony. the router, fyi, is
a piece of "shit". That's a rough translation of what I found about it.
I've regressed to asterisk 1.6.0-beta9 again and I had to reconfigure jackd
a bit to work. Byut - as before when I used it - the
The extra information does help. Unfortunately the Samsung 3010 is not
available here and the information I could find about it was all in German,
which I don't know fluently enough to read. I also run the risk of assuming
standards that are in fact different in different countries.
The only DSL
Jonathan Thurman writes:
> Sorry, I am relatively new to the Asterisk project and probably don't
> fully understand how the release cycle for this project works. Are you
> saying that the minor releases are only for security bugs?
Minor releases aren't only for security bugs, in general. This
pa
On Fri, Jul 03, 2009 at 02:38:46PM -0400, Jerry Geis wrote:
> Description Alarms IRQ
> bpviol CRC4
> DAHDI_DUMMY/1 (source: Linux26) 1UNCONFIGUR 0
> 0 0
>
> Is there a way to configure dahdi_dummy so that status report
Hello gang,
We just got MaBell to turn on our callerid. I tested the
capability with a southwest bell box and a plain phone, so I know the line
is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a
TDM400P card. Here is my dahdi_cfg -vv output:
dahdi_cfg -
Hello!
I've installed asterisk and now it crashes on me, here are three core-dumps
plus a note, saying which command created them and all I could gather in
/var/log/asterisk.
http://juliencoder.de/ast_debug.tar.bz2
the asterisk version running now is:
1.6.2.0.beta3
But it seems, that mIS
Description Alarms IRQ
bpviol CRC4
DAHDI_DUMMY/1 (source: Linux26) 1UNCONFIGUR 0
0 0
Is there a way to configure dahdi_dummy so that status reports OK
instead of unconfigured.
Jerry
_
Hello!
Thanks Wilton! You pointed me to the fact, that my initial post was a bit
unspecific. so here's the setup (hopefully acurately detailed).
The phone company promsed a full DSL+phone package. So there's the phone
jack in the wall, which is connected to a box, which I called router. Thi
At 12:33 AM 7/3/2009, you wrote:
>On Thu, Jul 02, 2009 at 08:15:37PM -0700, Ira wrote:
> > my DAHDI lines work with incoming calls, but I can't make an outgoing
> > call on a DAHDI line.
> >
> > The simplest line is:
> > exten=> s,1, dial(DAHDI/1)
>
>Extra space?
Not in extensions.conf that I
Un-top-posting
On Fri, 3 Jul 2009, selmak se wrote:
>>> I was trying to trigger an action (using AGI) when the recipient of a
>>> call (B number) answers his phone. (Actually the goal is to send a
>>> signal to an external server when the call starts)
On Fri, 3 Jul 2009, Philipp Kempgen wrote:
I may not be the best one to answer this because it sound like European
ISDN, but there seem to be some basic issues that might be relevant
>a Samsung router with analog and ISDN ports.
I'm not sure what this has to do with the voice side of things. An ISDN
port on a router would generally impl
I am trying to connect Asterisk and Openfire together, but it's not yet
working completely...
I don't know for sure if my manager.conf-file is set correctly. I use
this manager.conf file just to let Openfire talk to Asterisk...
[general]
displaysystemname = yes
enabled = yes
webenabled = yes (i
Trigger an action when B number answers the call.txt
Description: Attachment: Trigger an action when B number answers the call.txt
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selmak se schrieb:
>
>
Plain text?
http://lists.digium.com/pipermail/asterisk-users/2009-July/234274.html
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-aster
These may just be WAG's, but they are worth a try. Put exten =>
s,n,goto(wherever|s|1) at the end of the macro to make it go to the next
live place. As for starting the AGI when the caller answers, you can't
really do that; you either have to start the AGI before answer or run it as
a DeadAGI on
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Recently, I've been having issues with the URIs returned from e.164.org and
toll free calls. It seems that the URIs that are returned from ENUMQUERY and
ENUMRESULT are no longer the proper numbering schemes that the poviders use.
I've been using the following [enum] template in my outbound route
Chan_mobile supports SMS with a limited number of phones
On Fri, Jul 3, 2009 at 4:14 PM, Steve Totaro wrote:
> Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk
> supports SMS over GSM modem.
>
> I know chan_mobile had SMS in the future at one point but have not
> revisited th
abdelkader schrieb:
> What is the maximum number of simultaneous calls supported by asterisk.
42.
SCNR.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-
Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk
supports SMS over GSM modem.
I know chan_mobile had SMS in the future at one point but have not
revisited the project since.
"America Movil's MVNO TracFone Wireless quietly unveiled a prepaid,
nationwide unlimited offering for $
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digium card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients ar
That is the idea :) .
I'll search the web for more info.
On Fri, Jul 3, 2009 at 9:09 AM, Tzafrir Cohen wrote:
> On Fri, Jul 03, 2009 at 08:49:41AM -0400, Carlos Ruiz Diaz wrote:
> > Thank you for your help.
> >
> > Apparently I will rollback to AMI interface. I found a project named
> > Asterisk.
On Fri, Jul 03, 2009 at 08:49:41AM -0400, Carlos Ruiz Diaz wrote:
> Thank you for your help.
>
> Apparently I will rollback to AMI interface. I found a project named
> Asterisk.NET that interface AMI and my mono-C# application. I will be
> working in a solution.
>
> BTW, It will be truly interest
Thank you for your help.
Apparently I will rollback to AMI interface. I found a project named
Asterisk.NET that interface AMI and my mono-C# application. I will be
working in a solution.
BTW, It will be truly interesting the possibility of writing dialplans in C#
having all of the advantages that
Hello everyone!
I'm sorry I can't be more specific. So here's the setup:
a Samsung router with analog and ISDN ports. the phone company says the
outgoing line is analog landline, but I'm sure it's some VOIP.
so connected to the ISDN port of the router is a Fritz AVM card, used with
mISDN.
On Fri, Jul 3, 2009 at 12:36 PM, Tim Panton wrote:
> I'd try adding
> transfer=no
> in the B iax.conf
This does not help, I still have some ghost calls in B
a16-in1*CLI> core show channels
Channel Location State Application(Data)
IAX2/a16-in1-sangoma (None)
On Fri, 2009-07-03 at 11:58 +0100, Mike wrote:
> tempest:~# lspci
> 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
> interface
I don't think this is you TDM-card...
This is mine :
04:05.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
Subsystem: Digium, Inc.
search the mailing list, this question has been asked and answered several
times.
But it's all dependent on hardware, codecs, bandwidth.
If you mix the right technologies there is no limit to how many calls you
could handle, you just have to do it in the right way with multiple servers
obviously.
Depends on what you want to do and what your server platform is like.
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
abdelkader wrote:
> Hello,
>
> What is the maximum number of simultaneous calls supported by asterisk.
>
> thks
> ---
- "abdelkader" wrote:
>
> Hello,
>
> What is the maximum number of simultaneous calls supported by asterisk.
>
> thks
>
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Hello,
What is the maximum number of simultaneous calls supported by asterisk.
thks
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Folks,
I have a Xen Asterisk VM with a TDM400 card. When I try to run
dahdi_cfg, I get:
tempest:~# dahdi_cfg -vvv
DAHDI Tools Version - 2.2.0
DAHDI Version: 2.2.0
Echo Canceller(s):
Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: none) (
Hi,
is there a SIP/IAX equivalent of the ISDN-TON to find out what Type of
number (national, international, private) the submitted callerid is?
Can i somehow modify the callerid from a peer inside the peer-definition, ie
callerid="Gizmo5" <013${callerid(num:1)}>? I know this can be done via a
con
On Fri, Jul 3, 2009 at 12:36 PM, Tim Panton wrote:
> iax2 show netstats
The show netstats gives:
a16-in1*CLI> iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO
This book is a good place to start:
http://www.packtpub.com/article/asterisk-pstn-gateway-for-openser
G
On Thu, Jul 2, 2009 at 12:28 PM, Wesley Haut wrote:
> Are there any good tutorials or overviews on a basic setup using a SIP
> router in conjunction with Asterisk? I would love to get a proo
On Fri, Jul 3, 2009 at 12:36 PM, Tim Panton wrote:
>
> I'd try adding
>
> transfer=no
>
> in the B iax.conf
>
> I'm guessing the box in the middle (B) is somehow transferring itself out of
> the call
> but retaining a ghost call entry.
>
> It would be interesting to know what state those ghost call
From: Tzafrir Cohen
Sent: Friday, July 03, 2009 12:41 AM
>On Fri, Jul 03, 2009 at 12:30:07AM -0700, Trevor Hammonds wrote:
>> From: Tzafrir Cohen
>> Sent: Thursday, July 02, 2009 11:47 PM
>>
>> On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote:
>>
>> >> "Anyhow, on the Blackberry,
Hi Jonas, Hi Jeremy,
Indeed, typical examples for OpenSER are Load Balancing of clusters of
Asterisk servers (see here a nice tutorial
http://www.opensips.org/Resources/DocsTutLoadbalancing on that)
or fronting for security, protocol exchange (TCP/TLS), HA concerns, etc.
Regards,
Bogdan
PS: a
On Fri, Jul 03, 2009 at 12:30:07AM -0700, Trevor Hammonds wrote:
> From: Tzafrir Cohen
> Sent: Thursday, July 02, 2009 11:47 PM
>
> On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote:
>
> >> "Anyhow, on the Blackberry, when you hold down the Alt key and press the
> >> alpha character
On Thu, Jul 02, 2009 at 09:02:59PM -0700, Steve Edwards wrote:
> Access through the CLI (via -r -x) is problematic because you are subject
> to the whims of another process changing verbosity levels and you may have
> to sift through a lot of cruft as well as the overhead of invoking a
> proces
On Thu, Jul 02, 2009 at 08:15:37PM -0700, Ira wrote:
> I finally decided it's been long enough using my ancient HP junker
> and I built a Atom 330 based machine to replace it. I've installed
> Centos 5, Dahdi and Asterisk 1.6.2.
>
> After a bit of struggles getting the 1.2 version files converte
From: Tzafrir Cohen
Sent: Thursday, July 02, 2009 11:47 PM
On Thu, Jul 02, 2009 at 09:02:05PM -0700, Trevor Hammonds wrote:
>> "Anyhow, on the Blackberry, when you hold down the Alt key and press the
>> alpha character, the device sends out the correct digit tone associated
with
>> that character
On 3 Jul 2009, at 07:18, Rajkumar S wrote:
Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (
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