On Monday 10 April 2006 11:59, [EMAIL PROTECTED] wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
sip:[EMAIL PROTECTED] (french)
Hi,
sip:[EMAIL PROTECTED] (french)
sip:[EMAIL PROTECTED] (music 60s)
No Sound or voice!
/amp_maintenance.jpg
http://broda.homelinux.org/temp/amp_no-maintenance.jpg
- --
Thomas
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On Monday 10 April 2006 13:49, [EMAIL PROTECTED] wrote:
Could you try again please?
--- Thomas Winter [EMAIL PROTECTED] a écrit :
On Monday 10 April 2006 11:59, [EMAIL PROTECTED]
wrote:
Dear User,
Anybody could dial these sip uri :
sip:[EMAIL PROTECTED] (french)
sip
)
best regards
Thomas
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- Asterik-A - gsm - Asterisk-B - ulaw -POTS phone B
Any idea how this can happened?
If additional information required please ask..
best regards
Thomas
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Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem with
spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile
patch from the same directory.
When starting asterisk I always get
Rob Terhaar wrote:
did you try to recompile the plugin?
yes, of course...
On 4/14/06, Thomas Artner [EMAIL PROTECTED] wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same problem
after a few hours of debugging it works now...
I got some version mixes of spandsp on my system...
sorry for the spam
tom
Thomas Artner wrote:
Hi!
After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax
doesnt work any more.
I've installed spandsp-0.0.2pre25 (the same
Hi!
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!
At this point the
Michael Collins wrote:
A few months ago I needed some help for the following issue:
.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the
call
.) the caller get lost at this point !!
Melcon Moraes wrote:
So, what version of spandsp are using afterall?
i am using spandsp-0.0.2pre25 now.
In the 0.0.3 package, there is no app_rxfax.c and no app_txfax.c. No
idea why thats missing there.
tom
[]'s
MM
-Original Message-
From: Thomas Artner [EMAIL PROTECTED
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.
Is there something that would speak against it?
cheers,
tom
Thomas Artner wrote:
Hi!
A few months ago
here's the reported issue: http://bugs.digium.com/view.php?id=6973
cheers,
tom
Thomas Artner wrote:
Hi!
I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make
or is the documentation not up to date?
best regards
Thomas
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suggest if you
want to bypass that problem, add /s (or whatever extension) to the
register statement so you know for absolute sure that incoming calls on
the registration will go to the extension that you expect.
Aaron
On Wed, 19 Apr 2006, Thomas Winter wrote:
Hi,
the documentation
register with one number
and have it drop in on a totally different number in the context.
register = 44198:[EMAIL PROTECTED]/200
Aaron
On Wed, 19 Apr 2006, Thomas Winter wrote:
Hi,
[general]
context=Sip_in
register = 1234:[EMAIL PROTECTED]/s
s is the same, it still looks
Am Thursday 20 April 2006 01:21 schrieb tom:
Thomas Winter wrote:
I have done additional tests, because the documentation sample was not
100 % identical to my register command.
OK:
register = 44198:[EMAIL PROTECTED]/200
This jumps to 200, s is also working
NOT OK:
user:[EMAIL
://www.asternic.org). The way it is described
there, I could only make the Flash panel show that a queue 8in general)
received a call from a specific extension.
- --
Thomas Broda, Systemadministration Frankfurt
FIRSTGATE AG,Im MediaPark 5, 50670 Koeln
Telefon: +49 (0) 2 21 / 45 45-747
Telefax: +49 (0
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can hear the noise. And sometimes the call has to be hung
up, because the noise doesn't
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the Display Name)
for the initial INVITE to an SIP proxy.
If I use SET(CALLERID(name)=) the display-name is asterisk.
I want to have the SIP HEADER like this: FROM:
sip:CALLERID(number)@domain.tld
thanks
best regards
Thomas
some calls and disconnect the SIP-client every time.
best regards
Thomas
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Am Monday 24 April 2006 18:39 schrieb Doug Lytle:
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the Display
Name) for the initial INVITE to an SIP proxy.
If I use SET(CALLERID(name)=) the display-name is asterisk.
Just a guess, try:
SET
hmm.. does really nobody had such an issue before?
Thomas Artner wrote:
Hi!
I am using asterisk with two tdm400p cards.
Sometimes (one call out of ten), when a call comes in and is taken,
there is some terrible noise for a short time in the line (for about a
second).
Both partys can
Am Tuesday 25 April 2006 11:24 schrieb Olle E Johansson:
25 apr 2006 kl. 00.24 skrev Thomas Winter:
Am Monday 24 April 2006 18:39 schrieb Doug Lytle:
Thomas Winter wrote:
Hi,
I dont want to have in the SIP HEADER the CALLERID(name) (the
Display
Name) for the initial INVITE to an SIP
Am Wednesday 26 April 2006 20:43 schrieb Wai Wu:
If I download zaptel-1.2.5, do I still have to apply the
zaptel-1.2.5-patch?
no.
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I can not seem to configure to work with login. I thought it was pure
sip. It is unlocked. Can anyone help me
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Hi!
I've one card over from my last asterisk project.
The card is about 3 months old, a copy from the invoice for warranty is
available.
Location: Vienna, Austria.
If anyone is interested - send me a private mail.
cheers,
Tom
(i hope this mail is okay for this list)
I know I must be being daft, but is there a way to set which context the
queuing system uses when it dials the operators/agents?
By default it appears to use the default context.
I've looked through voip-info.org and can't find anything, someone
please put me out of my misery.
Alejandro Vargas wrote:
2006/5/23, Rene Nelson [EMAIL PROTECTED]:
I want to accept faxes via SIP/IAX2 (yes I've read the posts that it
isnt
reccomended). My PBX is 100% Virtual with the exception of one IAX
I made it with iaxmodem and hylafax. It can route fax to email
converting fax to
Alejandro Vargas wrote:
2006/5/23, Thomas Kenyon [EMAIL PROTECTED]:
Could you give me an example of the macro you use to convert outgoing
faxes from iaxmodem to emails?
With hylafax there are defualt files that works, but y changed some of
them.
I'm attaching the files /var/spool/fax/etc
Jon Scottorn wrote:
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a
number the script check to see if it is local if so, go out the ZAP
channel. If the ZAP channel is busy, go out the IAX channels,
Bruno de Assumpção Loureiro wrote:
Hi all,
How to integrate with Oracle database. I think it's possible with AGI,
it isn't?
Regards,
You will probably need to go down the odbc route for oracle.
Read the details on cdr_odbc.conf, extconfig.conf, res_config_odbc.conf
and res_odbc.conf on
Hi,
I have a digium T1 card installed on my Asterisk box. Protocol is PRI.
I am trying to setup so that the box can send and receive faxes. Being
able to receive faxes is a lot more important than being able to send.
I tried spandsp-0.0.2pre25, with proper app_rxfax.c file. But I am not
able to
Giorgio Incantalupo wrote:
Hi,
I'm trying to use Hylafax without a modem. Is it possible to use
t38modem to make Hylafax send and receive fax via Asterisk?
If yes, how? I'm searching on internet but still haven't found
anything useful.
Afaik, not yet, Steve Underwood who writes on the Dev
Bruno de Assumpção Loureiro wrote:
On 5/23/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
Bruno de Assumpção Loureiro wrote:
Hi all,
How to integrate with Oracle database. I think it's possible with AGI,
it isn't?
Regards,
You will probably need to go down the odbc route for oracle
Curt Shaffer wrote:
Sorry if this shows twice but it appears my first message was
quarantined because of my digital signature.
All,
I have been tasked with setting up video conferencing utilizing
asterisk. One of the requirements is a softset that has video
capabilities. Eyebeam
I run Asterisk 1.2.7 (upgraded from 1.0.x to 1.2.x etc.)
I notice that if I run the Management, the banner message I get is:
Asterisk Call Manager/1.0
According to voip-info I should be getting
Asterisk Call Manager/1.2
Is this difference significant?
I also get this with a machine that is in
bails wrote:
Hi all I fancied playing with SER and * on the same box. So i thought
i'd just change the default sip port for * in sip.conf
[general]
port = 5065 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
restarted * and
I have (more than 1) provider that I receive calls from using IAX, and I
have 2 IAX deskphones, all work fine except for some reason with 1
provider, when the call comes in, it doesn't match up with the
incomingcall context. (A bit worrying, since I don't want people to be
able to relay calls off
Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk.
For the second time now, I've had asterisk on a production machine
completely freeze (with no messages in any of the log files) and
eventually had to be kill -9'd.
The machine has a a TDM400 with 1xFXS and 3xFXO cards in
Doug Lytle wrote:
Thomas Kenyon wrote:
Is it neccesary to upgrade Zaptel at the same time as upgrading
asterisk.
I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons
and Sounds.
Doug
The problem with zaptel is that even if you can unload the modules and
reload them
Mike Hammett wrote:
I'm looking for an ATA\Voice Gateway that runs IAX and has several
ports (8 would be nice). I am looking to avoid devices that use the
same firmware as the ATCOM devices as I found them to be buggy (and a
PITA to find the proper update).
The Atcom devices use 2
Matt wrote:
In both cases, SIP/116 is on hook and available for calls.
The only thing different is in example one... before it rings
extention 116, it rings extention 101 for 5 seconds.
I know this sounds silly, but you didn't miss anything in the log
stating that the handset had become
Mimmus wrote:
Hi,
I just migrated my Asterisk installation from 1.2.1 to another server with
1.2.8. Among a lot of things, I copied the whole content of
/var/spool/asterisk/voicemail/default directory.
All is OK but now I'm not able to see MWI indication for new messages on all
my
Cory Andrews wrote:
I would not say the Cisco ATA has been replaced. Cisco continues to sell
the device, Linksys is a separate division entirely.
The ATA-186 has (2) FXS ports, each of which can be provisioned with their
own phone number. Each port also has built-in echo cancellation with
Erick Baum wrote:
The worst ongoing issue has been the echo and the really crappy
speakerphone. The customer is pretty much used to it now. But we're
slowly replacing them with Polycom's as new people come on and as
others just get fed up. Unfortunately one of the phones met it's
doom by
Mike Fedyk wrote:
First of all, I'm not knocking Sipura/Linksys. I have heard very good
things about their products.
I'm just wondering if they are the only quality shop on the market. I
know about the zoom 5801 where you can't dial out the FXO from SIP,
only from the FXS port. And I have
Mimmus wrote:
* Mimmus [EMAIL PROTECTED] [07-06-06 16:52]:
At first, I tried some chinese phones (AtCom) and they were
a disaster.
you talking ybout this phone?
http://iaxtalk.com/index.php?main_page=product_infoproducts_id=2
Yes
DV
I've been using some of the
Lachek Butalek wrote:
I've gotten nothing from http://bbs.atcom.cn on this so far. Perhaps
someone on the list has experience with this.
Is there a way to get MWI support for PA168V-based ATAs?
Afaik, none of the aredfox ATA firmware images support MWI, one reason
I've never bought one.
[EMAIL PROTECTED] wrote:
Hi,
Can anybody tell me that does asterisk have TAPI interface
sanchal
No, if you're a windows user, there is asttapi which uses the management
interface though.
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I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being sent.
exten =
Thomas Kenyon wrote:
I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being
I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running.
On the TDM400P, I have 1 FXS port and 3 FXO ports.
dmesg reveals:
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.6 Echo Canceller: KB1
PCI: Found IRQ 10 for device 01:01.0
PCI: Sharing IRQ 10 with 01:05.0
Nick Chalk wrote:
[EMAIL PROTECTED] wrote:
I've got speedtouch ones at home, here I've got
a Zoom one and a Dlink one I can try, It will be
a bit of a botch-job, atm. I'm using one of
those nice ones that plug into the front of an
NTE-5 (so I can punch the cables straight in).
An
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
I
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
no on-hook state in the whole setup.
Obviously If I just connect the input to a port on
James Harper wrote:
So... asterisk can't tell the difference between 's' for 'no extension
dialled', and when 's' was actually the name of the extension dialled...
is this the expected behaviour?
I surely hope so, you can refer to it as such in the extensions.conf as
well (with goto etc.)
amna saleem wrote:
hi !
i have installed asterisk-1.2.9.1
but am unable to run it
i am getting this error
[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined
symbol: ast_pthread_create
Jun 11 16:43:00
Doug Lytle wrote:
Thomas Kenyon wrote:
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
Does the connection use 2 screws
John Novack wrote:
Bob Chiodini wrote:
You might put 600 ohm/600 ohm matching transformer to isolate the
port and the amp. Should also maintain loop current if needed.
Bob...
FXO ports do not generate loop current, they detect loop current from
the Central office.
Think of an FXO as
Bob Chiodini wrote:
Thomas Kenyon wrote:
Mostly it uses the wrong impedance, and I know I can probably get an
impedance matching transformer, but I'm not allowed to spend any money
that I don't need to.
(Otherwise I'd have replaced the amp in the first place with one that
didn't fall from
Min Qiu wrote:
Hi all,
The firmware I used is pa168s_iax2_us_151011.bin.
My problem is the handset dial before I finished key in all
the numbers, no matter how fast I managed to press the keys.
It appeared it always dialed immediately, for example 011862,
when I actually ment to dial
Min Qiu wrote:
Arh... I did experence sound quality issues and I pointed
my finger toward my VoIP provider;-) Good to know. Can
you pass a pointer to where I can get 1.50?
Thanks a lot,
Min
Atcom support will send you one by email, or I can email you a copy.
Thomas Kenyon wrote:
Doug Lytle wrote:
Thomas Kenyon wrote:
Is it neccesary to upgrade Zaptel at the same time as upgrading
asterisk.
I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons
and Sounds.
Doug
The problem with zaptel is that even
Using Asterisk agents.
Not recognizing that an agent has made an outgoing call IS THE PROBLEM.
Only workaround I see is to take the agent out of queue on all outgoing
(and direct dialed incoming) calls and put him back in the queue at the
completion of the call. That seems too kloodgy.
Hence the
to an
agent if they are already on a call from the queue, but an incoming
call from another internal extension, or even a DID ought to be able
to get through.
Consider this a feature request?
Tom
On Oct 15, 2005, at 10:04 PM, J Thomas wrote:
One of my friends is facing
I've had an absoloutely fantastic run with the new KB1 patch currently
on mantis - http://bugs.digium.com/view.php?id=5520
The Digium guys are looking for feedback, please apply and test - If we
can get some positive feedback, it might make it into 1.2!
--Rob
Look at zaptel/zconfig.h and see what is uncommented.
[...]
I am using the KB1 echo canceller.
The 'new' echo canceller is MG2. Please test. This is _extremely_ good
on PRI, but it's been reported to have some issues on 2 wire circuits.
--Rob
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--Rob
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Has anyone had success using the SRV functionality in Linksys PAP2-NA's?
Regards,
David
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Hi, i found 2 bugs in asterisk 1.2.0rc1.
I using debian stable.
I start asterisk with:
/usr/sbin/asterisk -U thomas
or an different user,
Asterisk is starting.
Autodialing are Ignored.
(/var/spool/asterisk/outgoing).
Asterisk ignore to dial a Number / Extension, automaticlly.
When i start
When asterisk is setup to allow SIP users to send media end-to-end
(canreinvite=yes), can cdr info still be reliable, considering one of
the end-user devices could go down leaving the call open. This is
assuming you are using a third party pstn and not asterisk for pstn.
Does asterisk have any
Thanks for the information Matt!
Does asterisk store any SIP dialog cdr info in mysql like Call-ID
Cseq? With This info I could at least detect runaway calls and fake a
BYE to the pstn gateway with an external app.
regards,
David
On 11/23/05, Matt Riddell [EMAIL PROTECTED] wrote:
David Thomas
Kevin,
Is the CDR accounting done based on SIP signaling? If a UA is talking
(RTP) to a third party PSTN gateway, isn't it at risk if say the UA
loses power. How will asterisk know the call has ended if it is not
involved in the media path. The idea is this.. I want to use
canreinvite =yes to
Does asterisk fully support DNS SRV lookups yet, or does it still only
read the first SRV entry?
Info on the wiki looked quite old, so I thought I better ask.
regards
David
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Does asterisk have support for SIP session timers?
David
On 11/24/05, Olle E. Johansson [EMAIL PROTECTED] wrote:
Matt Riddell wrote:
Kevin P. Fleming wrote:
Matt Riddell wrote:
So how does Asterisk know that the media stream has been disconnected
between
the two remote hosts?
It
.
- --
Thomas
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You need to care about the _actual_ error, not the report there is an
error. The error is (usually) reported to the console. Reboot the
computer and type this:
dmesg -c /dev/null
modprobe ztdummy
dmesg
The output of the second dmesg will show you exactly what the error
message is.
Being that
-Original Message-
Thanks for the input Tony, but the instructions that Rob Thomas wrote
took care of my issue.
Thanks again to both of you!
You're welcome, Happy to help.
--Rob
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I use the Via PD-1 motherboards. They have 2 ethernet interfaces,
4 serial ports, 6 USB and a PCI slot. (And they have a fan 8)
Unfortunately, VIA can't supply the demand for these things at the
moment 8-(
http://www.viaembedded.com/product/epia_PD_spec.jsp?motherboardId=241
--Rob
It's a udev configuration. Read UDEV.txt, or, read the AMP wiki
http://aussievoip.com for a step-by-step.
--Rob
-Original Message-
From: [EMAIL PROTECTED] on behalf of Jachin Rupe
Sent: Wed 7/09/2005 9:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I asked my telco to release caller name on the PRI. Earlier they were
releasing only the phone number.
I still did not see the name, but only the number in caller id. Actually
I now see number twice. When I inquired with them this is the response I
got:
I ran a trace on your TG. I see
I'm hosting soft-switch.org now - Steve has said he doesn't want FTP, so it's
all http now. Feel free to update the wiki.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, 22 September 2005 11:50 AM
To: 'Asterisk Users
/lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No
such device
This is correct. Your card uses 'wctdm' or 'wcfxs' depending on what
version of asterisk you're using.
--Rob
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All the unknown device means
is that your lspci doesnt know what the card is. Thats
all. Nothing more.
--Rob
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryanto Rachmad
Sent: Tuesday, 4 October 2005 7:43
AM
To: asterisk-users@lists.digium.com
Subject:
One of my friends is facing this problems and I could not find any
solution to that. Hence this post.
In her Asterisk PBX, she has programmed about 10 agents, and strategy is
rrmemory. Everything works fine. When an agent has received an ACD call,
another call is not presented to him as long as
Setting incominglimit = 1 does not really solve the problem as I had
already mentioned. That practically takes away the call waiting and will
block all incoming calls including direct dialed calls. She does not
want that. Moreover, incominglimit is deprecated too.
-- jt
On Sat, 2005-10-15 at
Hi Asterisk Users,
finally SIP 302 forward/redirect is
fixed.
You can now sucessfully use Asterisk + Nikotel
alltogether.
With this, you can make outbound calls from
your Asterisk to a Nikotel 99XX or even Nikotel PSTN number which both
get redirected (since they are Nikotel VOIP
I am using ISDN 64k + VOIP (iLBC, G729, GSM codecs only!) + bulk traffic
(FTP, P2P, E-Mail, etc.).
I am using tc, HTB 3.6 + finer tuned wshaper script.
It works pretty well for me.
The callee never misses any VOIP packet from my side.
So I guess HTB + QOS works pretty well, even for VOIP.
I use
, too?. And how can i tell asterisk
to sent all none SIP-ip calls to the gatekeeper over h323?
thx in advanced.
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail: [EMAIL PROTECTED]
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--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon
This is the difference between the QuadBRI cards and a normal ZapHFC where
you use zaphfc drivers.
Thomas
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 4:25 PM
Subject: Re: [Asterisk-Users] Quadbri in NT Mode against PBX.
On Thu, 29
signalling = bri_net_ptmp
Chan_capi with AVM card doesn't support that.
Greetings
Thomas
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 30, 2004 5:15 PM
Subject: [Asterisk-Users] zaphfc hardware sound trouble
Hi,
I've been learning asterisk
/${EXTEN},60)
exten = 1235,2,Congestion
exten = 1235,102,Busy
exten = 1236,1,Dial(SIP/${EXTEN},60)
exten = 1236,2,Congestion
exten = 1236,102,Busy
[h323-gateway]
exten = _X.,1,Dial(H323/[EMAIL PROTECTED])
My h323 Gatekkeper accepts connections on 217.9.24.23.
any hints for me?
THX
--
Thomas Küpper
calls from our gnugk to asterisk?
any hints for me?
thx
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail: [EMAIL PROTECTED]
Homepage: http://www.01063telecom.de
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Thomas
Kuepper
Envoyé : jeudi 5 août 2004 13:31
À : [EMAIL PROTECTED]
Objet : [Asterisk-Users] h323 gnugk to h323 asterisk and then to
endpoint
hi,
we are using a voip h323 switch. the switch sends all caals to our
Gatekeeper (gnugk
the merge conflicts in it.
I can only recommend this, if you are not really going to use the parking
functionality.
It can happen, that merging + fixing res_features.c will actually break the
logic.
One the other way, I don't have any SIP hangup problem :)
Thomas
- Original Message -
From
in extension.conf, finds 069xx and routes the
call to the sip endpoint?
no i think, in my case all works fine. the call arrives at the gnugk. I
have the prefix set to 069 but gnugk didnt sent the call to asterisk.
where am i wrong?
thx,
thomas
Am 05.08.2004 um 15:14 schrieb eltorio:
Same because you
to the gatekeeper. here is my extensoin for the gateway. Why
das asterisk send all calls to the gatekeeper instead of to the
gateway?
[h323-gateway]
exten = _X.,1,Dial(H323/h323:[EMAIL PROTECTED])
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241
not. whats the Problem?
-- Executing Dial(H323/ip$217.9.21.6:2554/2969, SIP/0699073201)
in new stack
-- Called 0699073201
== No one is available to answer at this time
thx
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434
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