On Sat, 2004-09-25 at 09:49, Michael Bielicki wrote:
64bit it :)
[EMAIL PROTECTED] root]# cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 15
model : 5
model name : AMD Opteron(tm) Processor 244
Any idea to the number of channels your
I have calls coming in via SIP (a DID) and I want to forward them right back
out to my cell.
If I do it in one step,
(as if 2125551212 was the DID, and 202111 was my cell number)
exten = 2125551212,1,Dial(SIP/${PROVIDER}/1202111,60)
The call comes in via sip, my system sends the invite
I use Digium's Licensed Codec and I have no problems in routing calls to
either E1 or T1 interfaces.
But ...beware of the Pitfalls in using non-standard G729 Codecs.
I used a couple of sets before and here are the problems I found (I have not
used Daniels codec though):
1) Calls are too noisy and
Hi,
As I am the developer of DIAX
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
there is already iax softphone called diax
(http://www.laser.com/dante/diax/diax.html) that can be controlled over
bluetooth on some phones. The thing that is missing is to be able to use
cellular
Title: Message
HI,
Can anyone please
tell me
1) Where does
asterisk store the call detail records?
2) What is
thestructure of these call details records?
2)How to
access the call detail records by any external application?
Thanks in
advance
Regards,
Mayank
Hi all,
I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN
cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest
testing versions).
If I start asterisk I get: chan_capi.c:2635 load_module: CAPI not installed.
lsmod | grep capi
On Sat, 25 Sep 2004, Steve Underwood wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I am not a lawyer, nor even a US citizen. Talking to someone who is both
may be a good idea.
What is the relevance of being a US citizen? Copyright rules are largely
global.
There are two
find someone to host it in India or serbia and you can safely ignore it :)
On Sat, 25 Sep 2004 10:22:52 +0200 (CEST), Peter Svensson
[EMAIL PROTECTED] wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I am not a lawyer, nor even a US citizen.
Hello Bjoern,
thanks for this nice discussion;
we we dod have msn (4) although the telco company tells us that we
have pp isdn. This seems to be a little bit strange to me... Is there
any way to crosscheck the isdn configuration ?
And what about the active or passive isdn cards ?
I just want
On Sat, Sep 25, 2004 at 10:12:44AM +0200, Joost Kraaijeveld wrote:
Hi all,
I am trying to get my Debian Sarge to work with 2 Winbond W6692 chipset based ISDN
cards and Asterisk 1:0.9.1+1.0RC1-8. I have installed CAPI and chan_capi (all latest
testing versions).
If I start asterisk I
I am not an OnDo user. Please do not spam me.
- Original Message -
From: SeshKanuri [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 4:47 AM
Subject: [Asterisk-Users] Move Over Asterisk - Ondo is Here. -
Well maybe you should be a user. I offer much less than *, at only a
much greater cost :-)
I think this is a bit like advertising Windows XP on the Linux kernel
mailing list :-)
Regards,
Steve
Steve Totaro wrote:
I am not an OnDo user. Please do not spam me.
- Original Message -
Michael Bielicki wrote:
find someone to host it in India or serbia and you can safely ignore it :)
On Sat, 25 Sep 2004 10:22:52 +0200 (CEST), Peter Svensson
[EMAIL PROTECTED] wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I am not a lawyer,
Title: Message
HI,
Can anyone please
tell me
1) Where does
asterisk store the call detail records?
2) What is
thestructure of these call details records?
2)How to
access the call detail records by any external application?
Thanks in
advance
Regards,
Mayank
Is it April 1st already, where did the year go
Andy
On 25/09/2004 at 01:47 SeshKanuri wrote:
Dear Valued OnDO users,
OnDO PBX v1.3 now supports 100 concurrent calls
Brekeke is
On Sat, 25 Sep 2004, Steve Underwood wrote:
But the patches aren't a derived work. That is the value they have here.
There are an independant adjunct work.
According to most lawyers a patch _is_ a derived work in nearly all
circumstances. E.g. a novel based on the characters from a
Please can someone help me to install chan_capi on Mandrake 10. I get page
after page of errors and can not seem to find detailed install instructions
anywhere.
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I would like
asterisk to dial an extension or external number but for the call to only be
connected after the called party presses a key. Therefore been able to announce
the call to the called party before answering. I have had this working on
queued calls but want to incorporate this for
Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).
Config
FC2 running Asterisk 1.0.0, with the h323 compiled in and installed
James Bean wrote:
Sorry to post such a newb set of questions but I have been hammering
about trying to get Asterisk running on FC2 machine reading everything
available (I think that is what stuffed me, shouldn't have read it all
:-) ).
Config
FC2 running Asterisk 1.0.0, with the h323 compiled in
Or what is it that you meant in particular?
I'l bet he means 3rd party call control like in a traditional CTI
deployment ala Cisco ICM, Genesys or an oldie-but-goodie, IBM CallPath
DirectTalk.
(Net-net version)
Basically, a scratch-pad type area of ~2K that gets created/destroyed
with every call
Hi Joost,
the W6692 based cards do NOT have capi drivers. At least not with
isdn4linux, maybe it would work with the mISDN drivers.
I have a W6692 card laying around on my desk (thanks voidptr :) ),
a zaptel driver for that chipset is planned, but of course other
things are more important. ;)
On Saturday 25 September 2004 06:03, Peter Svensson wrote:
As an example, if I were to write a few more chapters to Gone With the
Wind those would be a derived work and, in countries signatories to one
of the two copyright treaties, the property of the original copyright
holders.
IANAL, but
Hi!
The first hurdle you must take is finding out what busy exactly means
for your SIP phones - do you allow only 1 call appearance, or 2, or ...
see the dialplan commands SetGroup, GetGroupCount etc. for this.
Note: Before this feature was added to Asterisk people used
outgoinglimit= and
Hi folks,
I'd like to encourage all of those friendly mirror maintainers to include
their link here in the appropriate place:
http://www.voip-info.org/wiki-Asterisk-mirrors
Cheers, Philipp
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[EMAIL PROTECTED]
I was really looking forward to Asterisk 1.0 et al, but it is a major
disappointment. I have never experienced any Asterisk release that was
interacting with Digium hardware so unreliably.
Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as
the call is being picked up at the
On Sat, Sep 25, 2004 at 11:18:23AM +0200, Thomas Niesel wrote:
Donno if zaphfc would be useable right now or in near future!?
Worked fine for me here with $20 card until entire Alcatel pbx locked up
and they blamed our line..
arkadi.
___
Title: Nachricht
Sorry, I cant help,
but I do have the exact same problem with compiling chan_capi module under RH
9.0.
Anybody any
Idea?
Thanks
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[EMAIL PROTECTED]
-- snip --
Had the patch been against the actual g729 libraries the case would have
been clear. Now, the patch is against asterisk to make it interoperate
with the g729 libarary and this may or may not be non-infringing. However,
the distribution of the g729 libraries themselves are almost
On Sat, 25 Sep 2004, Andrew Kohlsmith wrote:
IANAL, but those chapters would be yours. Adding them to Gone With the Wind
and distributing the resultant new book would be considered distributing a
derived work and fall into the gray area.
It's the same as fanfic; the characters and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello together,
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE minimum space requirements for
~ running asterisk to work
Am Sa, 2004-09-25 um 14.31 schrieb Arik Funke:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello together,
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE
On Saturday 25 September 2004 08:31, Arik Funke wrote:
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE minimum space requirements for
~ running asterisk to work as
On Saturday 25 September 2004 08:03, Benjamin on Asterisk Mailing Lists wrote:
Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as
the call is being picked up at the other end.
Disable callprogress and/or busydetect.
I have tried various X100P (original Digium) cards,
We've been using the CellSocket on asterisks in our lab and
it works well. They only problem we found was
DTMF performance from the local cell phone to asterisk has varied
depending on carrier and phone model.
/ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Sat, Sep 25, 2004 at 06:48:02AM +0200, Goran Dj. arranged a set of bits into the
following:
I tried to install chan_sccp (make; make install) but after that when
asterisk starting:
[chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242
ast_load_resource:
Hello,
I'm going to use * as SIP-H.323 proxy (codecs doesn't matter - only
pass through). I compile * (v1.0.0) without any problems as far as
H.323 stack (pwlib, etc). But when I'm trying execute asterisk -vvv I'm
getting error message:
[codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]:
Scenario,
I got some very good help earlier from Joseph getting me up and started
but I have a couple of small problems still.
Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4
Analog dialout line and Analog handset plugged in.
Problems:
1.
Incoming calls work and the phone
-Original Message-
From: Marco Nicolayevsky [mailto:[EMAIL PROTECTED]
Sent: Friday, September 24, 2004 11:45 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] agents and queues
How can i determine if there are any agents signed-in,
and if not, take them straight to voice mail with
a
I've have the main Astricon dev conference from 12PM to the end recorded
and posted at http://snipurl.com/astricon . Due to overloaded hotel
uplink (T1) there are some spots with no audio where the uplink droped
out for a few minutes.
-Brian
___
Title: Message
see: http://www.voip-info.org/wiki-Asterisk+billing
- Original Message -
From:
Mayank Mishra
To: [EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 2:10
PM
Subject: [Asterisk-Users] How to get Call
Details Records
HI,
Can anyone please
Incoming calls work and the phone rings and can be answered
no problems, (although I wouldn't mind being able to adjust
the ring but that's not important), I can't ring out, I just
get a busy signal and nothing comes up on the console. I am
pretty sure its just a simple line missing from
Hi
I'm try to get any variable (i.e.:CALLERID) on my agi script in perl.
Using the function get_variable(), the value is empty...
I read that the function don't work properly...
Please, ignore my terrible english (i'm from 'sao jose dos campos', brazil).
Thanks,
Ricardo Maia
On Sat, 2004-09-25 at 11:43 +0100, Nicolas Whitham wrote:
Please can someone help me to install chan_capi on Mandrake 10. I get page
after page of errors and can not seem to find detailed install instructions
anywhere.
So you phone the AA or RAC and say my car's stopped and nothing else,
On Sat, 25 Sep 2004 07:00:46 -0700, Brian [EMAIL PROTECTED] wrote:
we have a mirror for that at:
http://astricon.asterisk.pl/2004-09-recordings/index.php
--
Michael Bielicki
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sat, 25 Sep 2004 21:03:31 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
The problem of false hangups really needs to be fixed. A false hangup
is NEVER EVER acceptable in an office environment. On the other hand,
a call that doesn't hangup even if the remote party has
On Sat, 25 Sep 2004 15:18:27 +0200, Marcin Kwiatkowski
[EMAIL PROTECTED] wrote:
[codec_ilbc.so]Sep 25 15:15:43 WARNING[16384]: loader.c:248
ast_load_resource: /usr/lib/asterisk/modules/codec_ilbc.so: undefined
symbol: sqrt
Sep 25 15:15:43 WARNING[16384]: loader.c:429 load_modules: Loading
Hello,
I have a following setup:
IP phone (Cisco/Skinny) - * - NAT -- NAT - * - PSTN
Everything is perfect when i'm using it from right to left. From left to
right however, there is no voice, although the calls are being placed.
I played around with codeces but no change.
Does anybody know,
I cannot get incoming calls to sip phones behind a PIX to
work, outgoing is fine.
Asterisk (Public IP) Internet PIX (NAT) Sip Phones
I have tried no fixup protocol sip, I have punched a hole in
the Pix allowing anything from the Asterisk box into the network, still no
incoming.
el Flynn wrote:
Lenny Tropiano / asterisk.org Mailing list wrote:
These taken tonight (9/22/2004) at the Expo and Reception
Enjoy. http://photos.tropiano.org/gallery/astricon-2004
Lenny
Anyone knows if those Snom Keypad 220s are available, and where I might
be able to get my hands on a few?
I
On 25/09/2004 at 14:31 Arik Funke wrote:
Hello together,
I am setting up a communication server which should also act a
very-low-load PSTN gateway. I am usiing a AMD K6 200 running from a 500
MB usb memory stick. What is the ABSOLUTE minimum space requirements for
~ running asterisk to work as
On Sat, 25 Sep 2004 08:45:56 -0400, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
Disable callprogress and/or busydetect.
I wouldn't have posted without having tried that beforehand.
The problem persists with both busydetect and callprogress disabled.
Where are you located?
In Japan. Lines are
Ryan Courtnage wrote:
Hi all,
A while back, there was a short thread on using the FXS interface from a
Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO
interface on the TDM400P:
Primus -- DLink ATA FXS -- TDM400P FXO -- Asterisk
In that thread, a couple of people
I don't see anything posted here in extensions.conf to allow dialing out on
group 2.
You need something like this:
[outgoing]
exten = _9X.,1,Dial(Zap/g2/${EXTEN:1})
exten = _9X.,2,Congestion()
And add the context outgoing to those extensions that you allow to dial out
to the PSTN.
Lyle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andy Powell
Sent: 25 September 2004 16:27
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Absolutely minimal Asterisk PSTN gateway
On 25/09/2004 at 14:31 Arik Funke wrote:
Hello together,
I
I get this message at CLI.
what does it mean?
- shabanip
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Sat, 25 Sep 2004 11:41:12 -0300, Marconi Rivello
[EMAIL PROTECTED] wrote:
It is better than the call being disconnected in the middle of an
important discussion, and it may create the impression that the other
person slammed the phone on you if you were arguing or something like
that...
I have a customer that wants to try the
exact same thing next month. Unfortunately I dont have any advice for
you at this time. However, if the PIX doesnt end up working for you I
can tell you that Ive had excellent success with the INGATE product
line. (Both Firewall and Firewall
On Saturday 25 September 2004 11:28, Benjamin on Asterisk Mailing Lists wrote:
Disable callprogress and/or busydetect.
I wouldn't have posted without having tried that beforehand.
Fair enough, I saw that you'd written tried every option but a lot of people
don't actually mean that. :-)
Are any packets at all from the incoming call setup getting though the PIX?
In general, static NAT (plus access list), is required to enablean endpont with a global IP address to establish a connection to an endpoint behind the PIX with a private IP address.
Are you using static NAT and what
There's another legal side to all of this which we need to evaluate carefully.
Putting the list and Digium, at risk, by being in a position of having it used
to break the law.
Starting a few years ago ISPs became liable for harboring lawbreaking
customers, and ended up answering to the court.
steve szmidt wrote:
The law breaking would be trafficing in illegal copies of G729 with the intent
of breaking the law.
Clearly, there is ample evidence in the list archives that the members
of the list strove valiantly, in the face of greatly confusing and
generally burdensome IP laws, to
On Sat, 25 Sep 2004 12:41:03 -0400, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
Fair enough, I saw that you'd written tried every option but a lot of people
don't actually mean that. :-)
:-)
Lines are provided by NTT. The driver (wcfxo.o) has been
built with #define JAPAN uncommented.
I'm trying to hook up a non-PRI fractional T1 using a T400P port. The
Telco says that it is provisioned as AMI with SF (not ESF) and that they are
signalling by sending down a straight DS1 (I'm not sure what exactly that
means). They are also sending DNIS over these channels. I currently run
Hi! I have Asterisk up and running and have KPhone, IAXtel and FWD
connected to it and working. I have a Cisco 7960 with version 3 (App.
Load ID POS3-2-00) software. I have configured the 7960 correctly, I think;
I have set everything - name, shortname, auth.name and display name set
to 200.
I
On Sat, 25 Sep 2004 01:47:38 -0700, SeshKanuri [EMAIL PROTECTED] wrote:
Dear Valued OnDO users,
?
[snip]
For sales information, please contact us at
[EMAIL PROTECTED]
You have received this email from Brekeke Software Inc because you
registered to receive periodic news and updates
I've seen alot of posts lately on Queue and Agent functionality, and
alot of hacks to make them do different things that most call center
managers want.
In the sake of doing this one time, I'd like to develop a single list
of request so we can consolidate a feature request for the Queue/Agent
Chuck,
The first thing I would do is to upgrade the load to version 6 or
higher. I'm running the latest...version 7.2. (I'm very happy with it)
Are you using TFTP to load the configuration or manually configuring the
7960? I know it's a pain to setup TFTP just for a quick test. However,
it's
Is there a place to get the software load for the Cisco phone without
having a support contract? Buying the phone was costly enough, but now
needing to pay for the software to fix it is really poor!
Chuck Wegrzyn
Chad Brown wrote:
Chuck,
The first thing I would do is to upgrade the load to
Hi people,
I'm having some trouble with my analog phone on Zap/1 not ringing directly
when i call it from the PSTN via Zap/4, after about 3 rings the analog
phones rings. Now I understand that there is a slight delay of up to 3 rings
where is the tone detection, so no phantom calls will get
C Wegrzyn wrote:
Is there a place to get the software load for the Cisco phone without
having a support contract? Buying the phone was costly enough, but now
needing to pay for the software to fix it is really poor!
That's the Cisco way!!
They're not content to charge a premium price for their
SF framing is called d4 in the zaptel.conf. And use ami instead of b8zs.
If you want those changed, it will be basically a new circuit from your
telco!
You say that you have it running into a channel bank now. What type of
channel units are in the channel bank? That will tell us what type of
It works fine for me. I have a handful of Cisco 7960s
behind a PIX firewall and they register to a Asterisk server outside of the PIX
with no trouble at all. I didnt do anything special to the
PIX (i.e. no access list entries).
The tricks I found to make it work generally apply to any
Yes, that's tough. A couple things though...
1. To be fair...My 3.2 load did work against Asterisk. I just feel that
troubleshooting should begin with the latest bug fixes applied if
possible.
2. You may be able to contact Cisco technical support to get the latest
firmware / files. Before I put
In article [EMAIL PROTECTED],
Peter Svensson [EMAIL PROTECTED] wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
But the patches aren't a derived work. That is the value they have here.
There are an independant adjunct work.
According to most lawyers a patch _is_ a derived work in nearly
Hi.
I'm using asterisk as a PSTN - SIP gateway, so that you can call to any
of the 4 PSTN lines connected to the asterisk box from and dial your
number, and asterisk will dial out through one of the 4 sip accounts I
have on a SIP - PSTN provider. I think of something like this in the
You could start buy downloading my .iso (29mb bootable ) and use
that as a basis for your system. I've already modified it for a CF
card based system. Essentially it depends what sort
of interface to the PSTN you want. E1/T1 and analog should work
fine with my cd - but I've not built
it
On Sat, 2004-09-25 at 13:48, Brian Capouch wrote:
In the ideal world companies that treat their customers this way would
not be able to compete, but Cisco's de facto monopoly in the router
market allows them to treat their customers as if they were their inmates.
Not much choice until
All,
I am trying to do a dial to a cisco3660 endpoint. see the below
extensions.conf, sip.conf, and output to see my problem. Thanks in
advance for any input. In the debug look for the WARNING lines. thanks!
exten = 5149053538,1,Answer
exten = 5149053538,2,Wait,2
exten =
Thats Great news. Thanks for the
information.
What version of the PIX IOS you running?
Do you have sip fixup protocol enabled?
I have found a workaround, install onDo
sip server on a machine behind the PIX. The phones register to that, on the pix
port forward to the onDo sip
Aloha,
I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN.
I have an Asterisk box, RC2 with a for port FXS card providing
dialtone for a Norstar Key System.
I have it working so when you press a line key on the Norstar you get
dial tone from the Asterisk box. The user has
On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote:
Anyone else having the problems that Gary is reporting?
Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel
2.6) and i had to add a linux 26 at the end of the make line,
otherwise all kinds of weird things happened.
Also, in
Hi Everyone,
I've been playing around with Asterisk for awhile now, and keep having
this intermittent problem with my X100P... Here is my setup:
Linux Kernel 2.4.26
Wildcard TDM400P (One FXS port)
Wildcard X100P (One FXO port)
Running the 1.0 release of Asterisk and the Zaptel drivers
It seems
Does anyone know of a company that provides German DIDs (preferably Berlin)
and termination of calls to Germany at reasonable rates?
Thanks,
Eric
[EMAIL PROTECTED]
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If you have developed CGI, PHP or other synchronous web-based applications
that utilize the Asterisk manager interface, you know that they don't
scale well, since each invocation from the web requires a connection to
Asterisk and authentication there (thus putting a potentially large amount
of
from asterisk' point of view holding onto some sort of,a dn obtainign some
sort of uniq ID can be done easily via AGI and variableshowever, it
sounds like, what you're talking about is more of an app (with several
calls) and an resource too...
maybe not really certain.
--On Saturday,
Matt, I am tring to use cisco as a sip to pstn gw as well. are you using
an inbound sip dial-peer? or is not required? for inbound h323 calls its
not but i keep getting
Sep 25 15:47:23 WARNING[1087986608]: chan_sip.c:598 __sip_xmit: sip_xmit
of 0x81487dc (len 755) to 210.50.7.213 returned -1:
Regardless of whether or not you have licensed G.729 from SIPRO
independently of Digium, the distribution of the codec, linked against
Intel's proprietary IPP library, is clearly and totally in direct
violation of the terms of the GPL. There is no room for argument on this
issue.
We are
Hi,
if i understand german telco regulations right (even for a german that's
not an easy task...) then a provider may not assign a DID to a non-local
client. This would mean that a provider in Berlin may not assign a DID
to a client in Munich. So, assigning german DIDs to foreign clients
would
--On Saturday, September 25, 2004 23:28 +1000 James Bean
[EMAIL PROTECTED] wrote:
1.
Incoming calls work and the phone rings and can be answered no problems,
(although I wouldn't mind being able to adjust the ring but that's not
important), I can't ring out, I just get a busy signal and
Try www.sipgate.de . They have DID numbers available in 14 cities in
Germany.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Sat, 25 Sep 2004, Klaus-Peter Junghanns wrote:
Hi,
if i understand german telco regulations right (even for a german that's
not an
Hi,
Steve Underwood wrote:
Asterisk. I have been building and testing with the current * CVS code.
I still need to work through the national variants, and get some of the
them better tested. If you have the equipment ready to try MFC/R2 please
tell me how you get on.
I might have an R2 line
Chris,
I agree with your assessment of asterisk's queues. I took Robert's reply to
my original post, and came up with a way to tackle your first scenario (no
agents in queue=caller in limbo) with his idea of setting variables. My idea
deals with setting global variable states for each agent. I
Specs
FC2, Asterisk 1.0.0, Zaptel 1.0.0
TDM400P Port 1 FXS Port 4 FXO
Standard analogue handset plugged in with pstn line.
Problem:
When I go to dialout it drops numbers on the outgoing number.
Keys dialed from handset were
9 0418800185
I tried hitting the keys slowly as well as at my
Hi!
[fromcsr1]
exten= 800,1,AgentCallbackLogin(101|[EMAIL PROTECTED])
exten= 800,2,SetGlobalVar(GCSR1=on)
exten= 800,3,Hangup
determine which path the caller takes. The part that doesn't work is in the
[fromcsr] context. My SetGlobalVar(GCSR1=on) seems to have no effect,
therefore, making
Here is my resolution to the problem, I use AgentLogin vs
AgentCallBackLogin.
This is a long post, but I think it is very useful... :)
Call comes in via DID, queueable is a macro I wrote. ty_voice and
voice are
two sound files. The first one is used to play a Thank you for calling
XXX.
The
Klaus-Peter Junghanns schrieb:
Hi,
if i understand german telco regulations right (even for a german that's
not an easy task...) then a provider may not assign a DID to a non-local
Hi,
it's right, that german RegTP, the authority, who assigns number
ranges to telcos, now explicitely forbids to do
Hi All,
I consider the License fee charged by digium for G.729 as very reasonable,
and hope people agree and do nothing to jeopardize this project.
Right now I don't use G.729 at all, however if and when I do, I have no
reason
to seek an alternative to what Digium provides. At the very least I
Using X-Lite FWD soft phone, I can register, get to the 'demo' menu
extension, but that's it. Audio starts, then after a few seconds stops,
with packets still being passed.
Anyoen have any clues? Yes there are firewalls between here and there, yes
there is NAT at my end...What ports need
Try sipgate.de.
They have free DIDs in many german citys and their rate into Germany is
very affordable (aprx. $0.02 / min.)
Their website is in German only though.
Alfred.
Klaus-Peter Junghanns wrote:
Hi,
if i understand german telco regulations right (even for a german that's
not an
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