René Enskat [Teamware GmbH] schrieb:
Hi,
I want to try the skinny/sccp protocol.
Somebody can give me a working config for a cisco 7960 or 7970 ip phone?
Isit possible to forward a SIP extension to the skinny phones?
Coz i use normally a sip phone and i only want to forward this calls to
Guys, is app_followme going to be integrated into 1.2beta?
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
U need to set your digitmap.
Morel Mosolff wrote:
Hi,
sorry - I know that problem is not directly related to asterisk but mabe
someone can help anyway.
After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is
mostly not possible to dial numbers with leading zeros like
Euhm,
He was at von too, ask Kristian what a blast it would be.
Zoa.
Matt Darnell wrote:
Well that didn't take long!
He was a really nice guyI bet it would be a blast to go have a
beer with him.
We met him at the Internet Telephony Expo.
On 11/2/05, *Dean Collins* [EMAIL
Has anyone got distinctive ring detection working for PSTN lines in
Australia.
I am using the latest CVS and have got zapata.conf set up thus: but it
appears that the chan_zap modules is not going anywhere near that piece
of code and all it returns is the default 0,0,0
[channels]
context =
Please, inform us when you finish your script.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sherwood McGowan
Sent: 28. listopad 2005 12:32
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
i'm trying to debug the zaptel drivers on freebsd 4.x, and am trying to
isolate the problem. it's either a locking issue within the freebsd zaptel
drivers or the threading library used on freebsd (libc_r). in order to
isolate that it's not the threading library, i've used pritest from the
On 11/03/05 11:03 Dean Collins said the following:
Captain Crunch J
http://www.webcrunchers.com/crunch/
we had him down in KL last year for our HackInTheBox Security Conference,
and i must say the experience was less than optimal with mr draper.
--
Regards, /\_/\
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Viva,
Witch trunk are you able to establish ?
SIP trunk beetwen asterisk and PBX ?
Oh323 ?
Joo
Manuel Silva wrote:
| Hello!
|
| I have installed 2 servers, one with SER integrated with PostgreSQL
| (Fedora Core 3) and the other with Asterisk
Hello,
I've been waiting ASTERISK-B2BUA for asterisk-1.2
Regards
Harry
--- Rafael R. GV [EMAIL PROTECTED] a écrit :
try this: ASTERISK-B2BUA
http://lists.berlios.de/pipermail/b2bua-users/2005-November/000155.html
Features: full vovida's b2bua radius emulation,
extended radius
Hi.
We want to setup our own Ip-telephony service. Were sick of unstable
ip-telephony-providers. So how do I proceed on doing this? What hardware and
software do we need?
Thanks,
Arne Morten Johansen.
___
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Nope, never really found a satisfactory solution to this..
On 11/3/05, Tom Rymes [EMAIL PROTECTED] wrote:
Hi there. I noticed a post you made to asterisk-users backin June regarding problems you were having with Asteriskif your internet connection went down. I am having thesame problem here,
I am planning to connect my Asterisk PBX to one or two POTS
lines, and am wondering if it is better to use a TDM card for
this, or one or two SIP devices with FXO ports on them (such
as an SPA-3000, Grandstream 488).
I think it largely depends on where you're located and how much work has
On Tue, 1 Nov 2005, WideVOIP wrote:
Hello
as from 1.2beta2 it's not possible to build chan_capi
we get compile errors
use_ast_mutex_init_instead_of_pthread_mutex_init
if someone as any idea to correct this in the source code of chan_capi
Which version of chan_capi do you use?
The
Hello all,
I've just returned from a visit to a client site where their existing
incoming lines are in the form of 5 ISDN BRI connections (for 10 channels
total).
We have successfully deployed Asterisk boxes with 2 HFC-based cards in the
past, but I've no idea how well a standard PC will handle
I would suggest using a pair of 4-port cards. The interrupts alone
from 5 PCI cards would kill most boxes. There is also an octo-card,
but I have no personal experience of that.
The manufacturer:
http://www.junghanns.net/en/produkte.html
For an idea of UK prices:
Dear Asterisk users,
can you suggest me a free service where I can test my IAX trunks? Thank you.
Gabor
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Asterisk-Users@lists.digium.com
http://www.0xdecafbad.com/Free-VoIP-Providers.htmlhas a list of some free
providers
-Original Message-
From: Gabor Horvath[EMAIL PROTECTED]
Sent: 11/3/05 3:13:07 AM
To: Asterisk-Users listasterisk-users@lists.digium.com
Subject: [Asterisk-Users] IAX test service
Chris Bagnall wrote:
I've just returned from a visit to a client site where their existing
incoming lines are in the form of 5 ISDN BRI connections (for 10 channels
total).
We have successfully deployed Asterisk boxes with 2 HFC-based cards in the
past, but I've no idea how well a standard PC
I would suggest using a pair of 4-port cards. The interrupts
alone from 5 PCI cards would kill most boxes. There is also
an octo-card, but I have no personal experience of that.
Hmm... the price is something of an obstacle - given that single BRI cards
can be had for sub-£20, justifying £425
Steve Davies wrote:
I would suggest using a pair of 4-port cards. The interrupts alone
from 5 PCI cards would kill most boxes. There is also an octo-card,
You also have the BeroNet cards (http://www.beronet.com), exists in 4
and 8 ports. (BN4S0 - 4 S0 interface card)
They are almost
On Thu, 3 Nov 2005, Chris Bagnall wrote:
We have successfully deployed Asterisk boxes with 2 HFC-based cards in the
past, but I've no idea how well a standard PC will handle 5 or 6 cards -
i.e. every PCI slot has a BRI card in it.
Any thoughts from folks who've tried this in the past?
On 11/3/05, Chris Bagnall [EMAIL PROTECTED] wrote:
I would suggest using a pair of 4-port cards. The interrupts
alone from 5 PCI cards would kill most boxes. There is also
an octo-card, but I have no personal experience of that.
Hmm... the price is something of an obstacle - given that
Try o reupload the mysql database again to
see if that work?
Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Omar McKenzie
Sent: 03 November 2005 00:27
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
A2Billing
While we are in a process of moving our office, we use soft phones which
connect over WAN/VPN to our Asterisk box in the old office.
We use IAX2 softphones configured to use iLBC.
When we call out using the softphone, the bandwidth usage is at about 3
KB/s (in and out), quality is fine.
Sorry for this shameless hijack, is there a version of brisuff/zaphfc for 1.2 ?
Steve Davies wrote:
On 11/3/05, Chris Bagnall [EMAIL PROTECTED] wrote:
I would suggest using a pair of 4-port cards. The interrupts
alone from 5 PCI cards would kill most boxes. There is also
an octo-card, but I
I think for SIP the control channel can still go through the proxy while
the data is bridged natively allowing you to still account for the
call. I'm not sure of the details on how Asterisk does it.
MARK.
David Bandel wrote:
On 11/2/05, Mark Hulber [EMAIL PROTECTED] wrote:
I think this
On Wed, 2005-11-02 at 19:33 +, Faris Raouf wrote:
Please note, however, that somewhere in the wiki it suggests that you
modify the AVM driver code slightly. I found this stopped it compiling,
and that simply leaving the code as it is worked fine.
Then please add a note to that page on
I have a few lines of code that check the current date (20051103) for a
database entry in HOLIDAYS and if it is there go to the 'Night'
attendant for business closures on holidays.
exten = s,4,SubString(TODAY=${DATETIME},0,8)
exten = s,5,DBGet(INHOLIDAY=HOLIDAYS/${TODAY})
exten = s,6,GotoIf
Paul wrote:
-Original Message-
What information do you need on the 7960?
Paul
Info relating to the 7.5 firmware version and it failing to register. Thus
needing a reboot to fix:
Hi.
Which Cisco firmware are you using? There's a known problem with lost
of registration with
No. Not at this time. It was introduced long after the feature freeze
for 1.2, and I wouldn't ask Digium to consider it based on that fact.
On 11/3/05, Anton Krall [EMAIL PROTECTED] wrote:
Guys, is app_followme going to be integrated into 1.2beta?
On Wednesday 02 November 2005 19:29, Kevin Hanson wrote:
Morel Mosolff wrote:
Hi,
sorry - I know that problem is not directly related to asterisk but mabe
someone can help anyway.
After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it
is mostly not possible to dial
Hello people, actually I'm new on this new technology (for me). Well I'll
tell you what do I have got and what I like to do, so
if I made the mistake to post this new thread here, just forgive me.-
Let me intruduce myself and what I am working on.-
Well, I'm from Argentina and, actually I'm a
Please do tell! :-)
On Thu, Nov 03, 2005 at 11:01:51AM +0200, Zoa wrote:
Euhm,
He was at von too, ask Kristian what a blast it would be.
Zoa.
Matt Darnell wrote:
Well that didn't take long!
He was a really nice guyI bet it would be a blast to go have a
beer with him.
i dial on my phone to to 8091222
and convert it on asterisk to #00#7091222
But Cisco says 404
cisco peer=
!
dial-peer voice 22 pots
huntstop
preference 5
destination-pattern #00#..\*
translate-outgoing calling 1
direct-inward-dial
port 0:D
prefix 810
!
Try FWD.
Gabor Horvath wrote:
Dear Asterisk users,
can you suggest me a free service where I can test my IAX trunks? Thank you.
Gabor
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Asterisk-Users mailing list
Hello,
I'm new to the list so I hope I'm asking the question in the right
place. In our extensions.conf, we call an AGI script using the AGI command.
e.g.
exten = 11,1,Answer
exten = 11,2,Wait(0.5)
exten = 11,3,Playback(welcome1)
exten =
Our free UK numbers can forward to IAX:
http://www.esms.com/services_numbers_pure_free.php
Simon
On 11/3/05, Gabor Horvath [EMAIL PROTECTED] wrote:
Dear Asterisk users,
can you suggest me a free service where I can test my IAX trunks? Thank you.
Gabor
On Thursday 03 November 2005 02:50, Gary Li wrote:
Tested but no effect!
Yes but where did you put it? Please post your /etc/zapata.conf.
-A.
___
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Asterisk-Users mailing list
Nope - I saw your posts on it though. Very frustrating. I've had to
discontinue use of my TDM FXS ports until some solution is found.
On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said:
Did you ever find a solution for this problem? I have it on latest Beta 2
Bart
- Original
Geoff Manning wrote:
Info relating to the 7.5 firmware version and it failing to register. Thus
needing a reboot to fix:
I don't have any documentation, but I can tell you that the 7.5 image
caused me ALL sorts of headaches. I rolled it out to a few phones to
test, one being our
I've had issues with the FXO port on the spa3000 - banking apps could
not hear the DTMF. I've also had problems with phones hooked up to the
TDM FXS ports where banking apps hear DOUBLE dtmf digits.
The only mix that seems to work for me is SIP phones / or analog phones
hooked up to ATA's and TDM
Did you try relaxdtmf=no
Walt Reed wrote:
Nope - I saw your posts on it though. Very frustrating. I've had to
discontinue use of my TDM FXS ports until some solution is found.
On Wed, Nov 02, 2005 at 10:18:47AM -0800, Bart Fisher said:
Did you ever find a solution for this problem? I have
On Wed, 2005-11-02 at 14:16 -0600, Rich Adamson wrote:
That's odd. I just checked our meetme using two C7960's and an external
Zap (pstn) call, and all worked as expected. Using cvs-head from early
morning Nov 1 on fc3 with analog TDM04 card.
Iirc to recreate the delay issue you have to use a
On Thursday 03 November 2005 02:50, Gary Li wrote:
Tested but no effect!
Yes but where did you put it? Please post your /etc/zapata.conf.
And, did he restart asterisk (not a reload)?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Sam Tam wrote:
Try o reupload the mysql database again to see if that work?
Sam
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Omar
McKenzie
*Sent:* 03 November 2005 00:27
*To:* 'Asterisk
Patrick wrote:
On Wed, 2005-11-02 at 19:33 +, Faris Raouf wrote:
Please note, however, that somewhere in the wiki it suggests that you
modify the AVM driver code slightly. I found this stopped it compiling,
and that simply leaving the code as it is worked fine.
Then please add a note to
On Wed, 2005-11-02 at 16:45 -0500, BJ Weschke wrote:
I've had the same experiences with systems I've put in production. No
degradation in quality until the number of simultaneous calls gets
well over 100 on a dual CPU machine.
May I ask which Asterisk version you use, which zap/iax/sip mix
Probably by preference and peer type matching, try setting a new VoIP peer
for inbound calls from asterisk
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ivan Vershigora
Sent: Thursday, November 03, 2005 10:27 AM
To:
I've had issues with the FXO port on the spa3000 - banking apps could
not hear the DTMF. I've also had problems with phones hooked up to the
TDM FXS ports where banking apps hear DOUBLE dtmf digits.
The only mix that seems to work for me is SIP phones / or analog phones
hooked up to ATA's
We're using SIP exclusively. We do use the meetme features that have
enter/leave sounds and name announcement and we've taken alot of the
patches (putting the playback of conference-wide announcements) and
integrated them in even though those patches were not merged with the
CVS-HEAD tree from
On Wed, 2005-11-02 at 14:16 -0600, Rich Adamson wrote:
That's odd. I just checked our meetme using two C7960's and an external
Zap (pstn) call, and all worked as expected. Using cvs-head from early
morning Nov 1 on fc3 with analog TDM04 card.
Iirc to recreate the delay issue you have to
Hello All , Can someone point me to a full description of all
options allowed with the include statement ? Tia , JimL
--
+--+
| James W. Laferriere | SystemTechniques | Give me VMS |
| Network
Note this is on external calls to external applications Not Asterisk
DTMF detection. It's as though DTMF is distorted when going through a
TDM fxs port, or that it's being caught (too late) and then
retransmitted. Does * intercept outgoing dtmf?
I haven't found good docs that tell exactly
Walt Reed wrote:
Note this is on external calls to external applications Not Asterisk
DTMF detection. It's as though DTMF is distorted when going through a
TDM fxs port, or that it's being caught (too late) and then
retransmitted. Does * intercept outgoing dtmf?
For outgoing DTMF play
This is a very interesting thread. Could folks posting their experiences
please also post the country their experiences relate to?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons
___
On Thu, 2005-11-03 at 09:31 -0500, BJ Weschke wrote:
We're using SIP exclusively. We do use the meetme features that have
enter/leave sounds and name announcement and we've taken alot of the
patches (putting the playback of conference-wide announcements) and
integrated them in even though
I understand. Are there or is there any other queueing application
for Asterisk that is more efficient than the out of the box Queue
application?
Thanks,
Waldo
On Nov 2, 2005, at 8:32 PM, Kevin P. Fleming wrote:
Waldo Rubinstein wrote:
Is this a feature/problem because I use
Eric Lyons wrote:
The basic function is to take an incoming DNIS/exten on one port, look
it up in the db, then dial out to another number on another port.
This is just basic dialplan work... why you would need a custom application?
___
--Bandwidth
Anton Krall wrote:
So this feature will be disabled for now? :(
What 'feature'? It was a warning message that I added and then
removed... there is no feature involved.
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Asterisk-Users
Yes, there is -T but it doesn't timestamp everything. All that the OP
posted would not be timestamped.
Kevin
Jorge Merlino wrote:
There is the -T option when running the CLI but I think it only works in 1.2
-- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in new
stack
-- Called
Try opening up iax.conf and where the softphone's entry is, set
disallow=all and allow=ilbc
Tomasz Chmielewski wrote:
While we are in a process of moving our office, we use soft phones
which connect over WAN/VPN to our Asterisk box in the old office.
We use IAX2 softphones configured to use
Has anyone run into this problem yet?
-Daniel
On 9/9/05, Daniel Corbe [EMAIL PROTECTED] wrote:
I'm getting a FLOOD of these types of messages on my MAC OS X box:
Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread:
Request to schedule in the past?!?!
Sep 9 14:46:37
I succed in configuring hylafax and iaxmodem, everything is OK now.
The only problem was the following:
if I start manually, as root,
/usr/local/bin/iaxmodem ttyIAX
/usr/local/sbin/faxgetty ttyIAX
and then start hylafax server everything is OK
If I try to add to /etc/inittab
Shame, looks like a good app
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|BJ Weschke
|Sent: Thursday, November 03, 2005 7:03 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] app_followme
|
| No. Not
Chris Bagnall wrote:
This is a very interesting thread. Could folks posting their experiences
please also post the country their experiences relate to?
We've had very good experience with the SPA-3000 in the UK since the
last version of the firmware sorted out local impedance settings
Mmhh maybe I'm not making myself clear... What I meant was, seems either I
have something misconfigured or this function/app in features.conf is not
working since I left what the default samples from 1.2beta2 but when pushing
the key combination, nothing happens..
Any clues how to debug this?
Does anyone know if it is possible to configure Asterisk in such a way
that it will ignore RFC2833 DTMF signals received from a SIP peer?
I am using Broadvoice for some DIDs at the moment and their system has
a tendency to mis-interpret DTMF digits, especially ones dialed from
mobile or office
Does anyone know if Asterisk supports any Voice recognition software
or is there a third
party out that has one available for Asterisk.
What I want to do with Voice recognition.
When some calls my * IVR instead of the caller spelling the name via
the buttons I want the user to be able to say the
On Thursday 03 November 2005 13:36, Chris Bagnall wrote:
I would suggest using a pair of 4-port cards. The interrupts
alone from 5 PCI cards would kill most boxes. There is also
an octo-card, but I have no personal experience of that.
Hmm... the price is something of an obstacle - given
My problem is slightly different as there is 2 T1 Ports involved - With a T1
test set I can clearly hear two tones sent quickly with each outside caller
press. I assume one of the tones is the actual audio passing thru the
connection and the other generated by the T1 card itself.If I make
Hi,
I have one host with fixed ip and two hosts with dynamic ip. These dynamic hosts should connect to the fixed ip host to register, so fixed and dynamic host can call each other without problem.
My question is, how to let dynamic hosts can also call each other? (use iax2)
Thanks for the
New problem..
now iaxmodem is up, started via inittab.
but it does not work...
Could it be a user problem ?
Currently I have:
I should add that I am using ulaw (g.711u) for all calls.
-RustyOn 11/3/05, Rusty Dekema [EMAIL PROTECTED] wrote:
Does anyone know if it is possible to configure Asterisk in such a way
that it will ignore RFC2833 DTMF signals received from a SIP peer?
I am using Broadvoice for some DIDs at the
Anton Krall wrote:
Mmhh maybe I'm not making myself clear... What I meant was, seems either I
have something misconfigured or this function/app in features.conf is not
working since I left what the default samples from 1.2beta2 but when pushing
the key combination, nothing happens..
Any clues
Hi all
I configured asterisk and webmin.i dont know how to
integrate webmin with asterisk and how to access
asterisk
through webmin.pls do the needful.
regards
ramakrishnan.n
__
Yahoo! FareChase: Search multiple travel sites in one click.
Chris Wade wrote:
Use 'timestamp=yes' in asterisk.conf instead of -T.
-T only affects messages generated by THIS connection (ie asterisk -RT
generated messages... not server generated messages.
'timestamp=yes' affects all messages generated.
And after adding timestamp=yes to
Hello All , Been looking at the timed allow functionality of the
'include =' statements . Without docs on the functionality I am plain
guessing about the syntax format .
I am trying to Allow a context the ability to dial out of my system at
a
time
Kevin Bockman wrote:
Yes, there is -T but it doesn't timestamp everything. All that the OP
posted would not be timestamped.
Kevin
Jorge Merlino wrote:
There is the -T option when running the CLI but I think it only works
in 1.2
-- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in
I have some issues with sending some faxes using spandsp (receiving
faxes is generally OK).
I noticed new versions of Spandsp come out every month or two, but they
don't contain a changelog (they do, but it's outdated).
Does anyone know if one can read anywhare what changed in Spandsp?
--
Thanks for the suggestion have tried
this already still having problems
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam
Sent: Thursday, November 03, 2005
6:54 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Hi all!!!
I have an asterisk compiled and started in one computer hereat home, so I create 2 sip useres that request autentication to the asterisk using X-Lite..
The useers are log in all right, but when i try to have a call between they, it not work...
I set the context as siptest, so what do
Hmm, yes, lots of comments about this problem. Doesn't seem like there's a near-term solution, which makes realtime extensions (at
least) rather unscalable.
I think I'll hack at app_addon_mysql.c and make my extensions.conf like:
[incoming]
exten=_.,1,LookupStuff(${EXTEN})
On Thu, Nov 03, 2005 at 10:13:34AM -0600, Anton Krall exclaimed:
Mmhh maybe I'm not making myself clear... What I meant was, seems either I
have something misconfigured or this function/app in features.conf is not
working since I left what the default samples from 1.2beta2 but when pushing
the
This is an old issue on which you can seach and find info. Some info
indicates that you need a timing source such as a zaptel card or
ztdummy. Other suggests that if you are using native music on hold and
mpg123 is in the path you might run into this error.
MARK.
Daniel Corbe wrote:
Has
Hi
I
was able to resolve issue of login on, updated database connection file in file
defines.php under web folder
However experience new problem. When login
to site gets blank page.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Omar McKenzie
Sent: Thursday,
OK,
From a provider: How do you think that you
re going to build your connection to the Outside pstn world? Do you have
a regular telephone line, or you are gong to connect yourself to a carrier through
an IP Trunk?
This is your first point.
You have already an Asterisk box, I
As I understand it, you can initiate the call by having one of the
dynamic endpoints call the other through the fixed ip host and then the
fixed host can allow the two endpoints to create a native bridge.
Otherwise, I think you'll have to somehow cache the registration at the
dynamics hosts and
It probably makes no difference to your problem but it's canreinvite
not canreinvete. You'll want to include dialout extensions in
[siptest]. For instance, maybe include your default context.
MARK.
Wagner Nunes wrote:
Hi all!!!
I have an asterisk compiled and started in one computer here
Frankly, I think this may be happening to me too. It's still a zap to
zap channel problem.
On Thu, Nov 03, 2005 at 08:27:59AM -0800, Bart Fisher said:
My problem is slightly different as there is 2 T1 Ports involved - With a
T1 test set I can clearly hear two tones sent quickly with each
Well, I hope many people feel that way about it. :-)
The best thing to do at this point is to download and test the betas
of 1.2 right now so we can get 1.2 released and we can move on to fun
things like app_followme post 1.2.
On 11/3/05, Anton Krall [EMAIL PROTECTED] wrote:
Shame, looks like
We've merged in 3599 and 4252 against a version of HEAD from around
the April timeframe of this year.
On 11/3/05, Patrick [EMAIL PROTECTED] wrote:
On Thu, 2005-11-03 at 09:31 -0500, BJ Weschke wrote:
We're using SIP exclusively. We do use the meetme features that have
enter/leave sounds
Hi all
my system 1:
celeron 1.2GHz + intel 810e (asus TUW-LA) + 256MB SDRAM
onboard vga (intel 810e chipset)
RTL8100 NIC
debian sarge 3.1r0a / kernel 2.6.8-2-686
asterisk / libpri / zaptel from CVS HEAD @ 2005-10-24
system 2:
pentium II 533MHz + intel 810e (dfi PW35-E) + 256MB SDRAM
onboard vga
OK, then...
I posted on the Bugs Web Site and markster said: This is a technical
support issue. Please pursue through Digium tech support
([EMAIL PROTECTED]) and contact me if you have any issues., Hmmm...
So I have written support - still waiting for answer - If I hear anything
I'll let
Erik wrote:
Sorry for this shameless hijack, is there a version of brisuff/zaphfc for 1.2 ?
Steve Davies wrote:
bad boy :-) there is none as of 'yet'. I guess Junghanns will deliver
one as soon as 1.2 is stable.
You could try the beronet-way (mISDN and chan_capi).
Cheers
Guess you've already seen the documentation on the wiki?
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime
Sorry if that's redundant to you but I think it answers the questions
you've posed to the list lately :)
Moj
PS - The inclusion of this dial-out context into your main context
Hi has anyone used MFCR2 using Unicall? I need to use the
protocol_variant=fx but Asterisk crashes saying that there isn'ty such
a module, though it appears as an option in the configuration file.
Does anyone know why it isn't working?
Jesus Mogollon
The useers are log in all right, but when i try to have a call between they,
it not work...
Read this
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
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Using cvs head downloaded as of just a few minutes ago..
chan_agent.c: In function `action_agents':
chan_agent.c:1446: warning: long int format, time_t arg (arg 7)
chan_agent.c: In function `__login_exec':
chan_agent.c:1684: syntax error before `char'
chan_agent.c:1701: `agent_goodbye'
Compiled fine here. What version of GCC are you using?
On 11/3/05, Matt Hess [EMAIL PROTECTED] wrote:
Using cvs head downloaded as of just a few minutes ago..
chan_agent.c: In function `action_agents':
chan_agent.c:1446: warning: long int format, time_t arg (arg 7)
chan_agent.c: In function
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