Hi,
I've found approximate same pricing for both. Sipura 2100 seems to have more
features...
What are differences between those two ?What about their reliability
(specially regarding fact, that they deal with analog phones) ?
Thanks in advance,
regards,
Rob.
__
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Po
But what happens if the H.323 device is on a dyamic IP? Do I need a
gatekeeper?
Thanks,
Humberto
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Adi Linden
Enviada em: Thursday, January 06, 2005 12:58 AM
Para: Asterisk Users Mailing List - Non-Commercial Dis
Wed, 5 Jan 2005 19:51:05 -0500 Noah Miller wrote:
> Hi All -
>
> I've got a load of Polycom phones, and for the most part, I think
> they're great, but one thing that is bugging the heck out of me (and my
> users) is the "on-hold" feature. When you're on a call, and another
> one comes in, it
hi,
i am using Asterisk CVS-05/31/04.
i have the problem that sip clients can make calls over asterisk
without registering befor. the xlite is not loged in with any
username/secret bit still can make calls over asterisk.
how can that be?
thx for help.
thomas
___
hi all.
was having problems with my phonecell connected to
wildcard fxo port. i get problems with detecting DTMF.
i have tried relaxDTMF but to no avail. i have asked
this before but would like possible causes. is it to
do with echo? problems with the GSM network? haven't
updated my asterisk for
Hi,
i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0) support for ip-trunking (HG3550).
So what if you have the following setup.
[OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP].
Am i right when i suppose t
Hey Dan!!
Give us a clue as to what hardware/setup & network provider you have there,
and we might be able to help :)
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 06 January 2005 06:56
To: asterisk-users@lists.digium.com
S
Thomas Küpper wrote:
hi,
i am using Asterisk CVS-05/31/04.
i have the problem that sip clients can make calls over asterisk
without registering befor. the xlite is not loged in with any
username/secret bit still can make calls over asterisk.
Not really an answer to your problem, but CVS versions
My suggestion is: don't use emerge for *. I am a big fan of Gentoo and
emerge, but some packages are just not mantained well. I had a lot of
strange problems when using emerge to install *. Just check it out from
CVS and compile manually. This also makes it easier to apply various pathes.
Remco
hi,
Am 06.01.2005 um 11:23 schrieb Ronald Wiplinger:
Thomas Küpper wrote:
hi,
i am using Asterisk CVS-05/31/04.
i have the problem that sip clients can make calls over asterisk
without registering befor. the xlite is not loged in with any
username/secret bit still can make calls over asterisk.
N
I tried for the first time to use mp3player(toto.mp3) in my
extensions.conf. This part
works perfectly.
But the sound quality is terrible. I tried to play the mp3 with mpg123
on my audio device,
plays ok. And I cannot found any information about the mp3player
application.
Do I need to encode t
I'm having exactly the same problem, I'm currently using * HEAD.
If anyone can help please let us know what is the issue.
Regards,
Humberto
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de rizwan
Enviada em: Thursday, January 06, 2005 4:34 AM
Para: asterisk-
Eric Bishop wrote:
Well it's clear now that this is not an isolated issue. Has anyone
been in touch with Digium about this issue? I have logged a support
issue with them, but thus far have not received a response. Anyone
had better luck with Digium support and the Compaq/HP G4 server
series?
On We
On Wed, 5 Jan 2005, Michael Graves wrote:
On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get this:
modprobe: Can't locate module wctdm
Any idea why?
my box
I really don't know why but after rebooting it just works (which solved
my problem but didn't get me any further as to why it did).
Thanks for all the tips!
On Thu, 6 Jan 2005, Niksa Baldun wrote:
My suggestion is: don't use emerge for *. I am a big fan of Gentoo and
emerge, but some packages ar
On Wednesday 05 January 2005 21:43, Steven Critchfield wrote:
> While browsing the Wiki, I found this bit of information. Maybe it will
> help you out some.
> http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts
Sadly it didn't do anything for me. Is there anyway to get any kind of status
On 6 Jan 2005, at 00:01, Michael Swan wrote:
Hi all,
I've struggled for several days trying to get a Digium TDM04B 4-port
wxfco card working on a Dell 1U PowerEdge 750 machine running
Fedora Core 1. I finally got a call back from Digium who indicated that
there is a fundamental conflict between the
Hello!
Am Donnerstag, 6. Januar 2005 00:42 schrieb Remco Barende:
> After emerging some updates this morning asterisk 1.0.3 fails to start
> root # Ouch ... error while writing audio data: : Broken pipe
I bet it's mpg123.
To figure out, what happened do the following:
emerge genlop
genlop -
Good day all
I have a pri card,e100
What I want to do is
If a fax comes in for number 1234567890 it should be e-mail to
[EMAIL PROTECTED]
If a fax comes in for number 0987654321 it should be e-mail to
[EMAIL PROTECTED]
ens
Can this be done and how
_
On January 6, 2005 06:55 am, Altus Snyman wrote:
> Good day all
> I have a pri card,e100
> What I want to do is
> If a fax comes in for number 1234567890 it should be e-mail to
> [EMAIL PROTECTED]
> If a fax comes in for number 0987654321 it should be e-mail to
> [EMAIL PROTECTED]
> ens
Yup it
Is a phonecell like a cellsocket? A device to convert a GSM phone into
a RJ11 phone connector? If it helps I had problems with DTMF (Asterisk
dialing through the cellsocket - outbound) and one of the problems I
had was the tx/rx gain settings in zapata.conf (I also had a problem
with asteri
Hi list!
I am going to install an intercom module for our home. The intercom can
cater for 3 doors where people can ring the doorbell and the unit can
also remotely open the door (Siedle system).
It would be nice however to know that you are opening the right door or at
which door the people ar
GIBERT Frédéric <[EMAIL PROTECTED]> wrote:
STLS4 is a 4 BRI ports card to connect to carrier.
STMD8 is a card to connect 8 ISDN Siemens phones (optiset)
STMD8 is not a board for Optisets. You have to use a SLMO/SLU board to register an Optiset/Optipoint500.
Do you Yahoo!?
The all-new My Yahoo
On Wed, 5 Jan 2005, Wiley Siler wrote:
> Telecom is complex but that does not mean that only a contractor can get
> it done. If I had money to pay a contractor, I would probably have had
> money to buy a boxed PBX in the first place. Cost effective has always
> been the greatest selling point to
Hello
Would it be possible to dail out to lett's say to 4 people with a .call
file and put them directly into a free meetme room.
Thanks
Sjaak
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I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did.
I setup an asterisk server on FC3 without any problems using with some
help from the wiki.
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3
It doesn't require a kernel compile, just a few conf file changes and a
di
and email-fax??
The other way around
On Thu, 2005-01-06 at 14:17, Andrew Kohlsmith wrote:
> On January 6, 2005 06:55 am, Altus Snyman wrote:
> > Good day all
> > I have a pri card,e100
> > What I want to do is
> > If a fax comes in for number 1234567890 it should be e-mail to
> > [EMAIL PROTECTED
Hi all
I've managed to get chan-oh323-0.6.5 working with asterisk-1.0.3
I've downloaded all the files from www.inaccessnetworks.com
pwlib + pwlib-janus patch
openh323 + openh323-janus patch
chan-oh323 0.6.5
Don't forget to apply the chan-oh323 patch to openh323 before c
Ronald Wiplinger wrote:
Me wrote:
10- Ran cat /proc/interrupts to make sure my card was not sharing an
interrupt with with any other hardware
Can you interprete it for my situation ?
CPU0 CPU10: 1469462555 1466917080
IO-APIC-edge timer
1: 153130 186604IO-A
Hi,
> Would it be possible to dail out to lett's say to 4 people
> with a .call
> file and put them directly into a free meetme room.
You could create an alias that dials 4 people at once, or you could generate
4 different call files. Seems to be easy enough..
Florian
___
Hi,
anyone has the above combination working?
On one * system I use a capi supported device which allows me to run
HylaFax in parallel with Asterisk. The other machine has a single ISDN
BRI HFC card, only.
Anyone who has a single HFC card and is using Asterisk and Hylafax
together like to share his
Altus Snyman wrote:
and email-fax??
The other way around
You can run a simple mail server on the * box to accept emails addressed
to the .fax domain (i.e. "[EMAIL PROTECTED]"). This presumes you are
able to forward the .fax domain from your main mail server to the * box.
Once you have the ema
Hi!
Is it possible to get four HFC-S based ISDN cards working in one server,
or are there going to be IRQ conflicts? Did anybody realize something
like that already (not with a QuadBRI card)?
Regards
Kai
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik
Does anyone have the sip subscribe working with the IP600 phones?
I saw a message about the Buddies, but I could not tell if it works.
Thanks.
--
respectfully, Joseph ===
-= ** =
___
Asterisk-Users mailing list
How do I fax a .tiff file with asterisk?
On Thu, 2005-01-06 at 15:13, Michael Welter wrote:
> Altus Snyman wrote:
> > and email-fax??
> > The other way around
> >
> >
> You can run a simple mail server on the * box to accept emails addressed
> to the .fax domain (i.e. "[EMAIL PROTECTED]"). Th
Hi All
I've bought a TDM400P and need some help with configuration. Can you tell me
what to do ?
I've tried to install and the message below has appeared:
[EMAIL PROTECTED] asterisk]# modprobe zaptel[EMAIL PROTECTED] asterisk]# modprobe
wcfxo/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o
Is the meeting still on for Saturday 1/8/05?
11:30am at 2375 University Av W STE120, Saint Paul.
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Altus Snyman wrote:
How do I fax a .tiff file with asterisk?
Use Steve Underwood's spandsp library and TxFax function in Asterisk.
See http://opencall.org
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
___
Is there a way to combine these lines into one?
exten => s,2,GotoIf($["${CALLERIDNUM:0:3}" = "800"]?s|108)
exten => s,3,GotoIf($["${CALLERIDNUM:0:3}" = "866"]?s|108)
exten => s,4,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|108)
exten => s,5,GotoIf($["${CALLERIDNUM:0:3}" = "888"]?s|108)
Thanks
--John
César Davi Ávila do Nascimento wrote:
> Hi All
>
> I've bought a TDM400P and need some help with configuration. Can you
> tell me what to do ?
>
> I've tried to install and the message below has appeared:
>
> [EMAIL PROTECTED] asterisk]# modprobe zaptel
> [EMAIL PROTECTED] asterisk]# modprobe w
Try this:
exten => s,2,GotoIf($["${CALLERIDNUM:0:3}" == "800"] ||
$["${CALLERIDNUM:0:3}" = "866"] || $["${CALLERIDNUM:0:3}" = "877"] ||
$["${CALLERIDNUM:0:3}" = "888"]?s|108)
Diego Aguirre
FWD# 459696
- Original Message -
From: "John Hill" <[EMAIL PROTECTED]>
To: "'Asterisk Users M
Arghh sure
Something like
Channel: ZAP/g1/xx/ZAP/g1/xxx/ZAP/g1/ etc
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: monitor
Extension: 601 <-- first meetme room
Priority: 1
ZAP/g1 is an ISDN30
But how do I know this 601 room is free
I've rooms 601 602 and 3,4,5,6,7,8,9 and so on
I li
On Thu, 2005-01-06 at 01:09 -0500, Noah Miller wrote:
> > I have these very phones and took me a while to figure this out myself.
> > The phone considers each line registration to be a line with a second
> > line. So, call line while someone is on a call and another instance
> > will appear below.
On Thu, 2005-01-06 at 11:15 -0300, CÃsar Davi Ãvila do Nascimento wrote:
> Hi All
>
> I've bought a TDM400P and need some help with configuration. Can you
> tell me what to do ?
>
> I've tried to install and the message below has appeared:
>
> [EMAIL PROTECTED] asterisk]# modprobe zaptel
> [E
Michael Welter wrote:
How do I fax a .tiff file with asterisk?
[..]
Use Steve Underwood's spandsp library and TxFax function in Asterisk.
See http://opencall.org
It is my understanding that TxFAX does not allow to callout and send a
fax, but to be used in Faxpickup only.
Do you have it working i
On Thu, 2005-01-06 at 12:36 +0100, Tais M. Hansen wrote:
> On Wednesday 05 January 2005 21:43, Steven Critchfield wrote:
> > While browsing the Wiki, I found this bit of information. Maybe it will
> > help you out some.
> > http://www.voip-info.org/wiki-Asterisk+TE410p+No+Interrupts
>
> Sadly it d
Yup, just to confirm, same here...
Anyone know of a fix?
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Humberto
Aicardi
Sent: 06 January 2005 11:00
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RES: [Asterisk-Users] asteri
Peer Oliver Schmidt wrote:
Michael Welter wrote:
How do I fax a .tiff file with asterisk?
[..]
Use Steve Underwood's spandsp library and TxFax function in Asterisk.
See http://opencall.org
It is my understanding that TxFAX does not allow to callout and send a
fax, but to be used in Faxpickup on
On Thu, 2005-01-06 at 12:15 +1100, Adam Goryachev wrote:
> Personally, I've never had a problem with Debian.
I'd second that. Just last month, I tried to install asterisk
with AVM proprietary CAPI drivers on FC3, and it didn't work.
The driver just could not be loaded. I then switched to Debian
S
On Thu, 6 Jan 2005 13:22:43 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:
> Hi list!
>
> I am going to install an intercom module for our home. The intercom can
> cater for 3 doors where people can ring the doorbell and the unit can
> also remotely open the door (Siedle system).
>
> It wo
Adam,
Tor sent this one a little while ago that looks really promising for
solving the problem.
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tor Setane
Sent: Thursday, January 06, 2005 2:09 AM
To: Noah Miller
Cc: Asterisk Users Mailing List -
On Thu, 2005-01-06 at 14:39 +0100, sjaak imap wrote:
> Arghh sure
>
> Something like
> Channel: ZAP/g1/xx/ZAP/g1/xxx/ZAP/g1/ etc
BAD idea, the fist person to answer will terminate the other calls here.
It is just like ringing multiple internal extensions until someone
answers.
> Ma
Hi,
I know this question may have been asked before (although the archives
don't seem to suggest it), but has anyone had any problems with Asterisk
accepting a PIN number for a conference room.
At this point in time I have established the conference definition in
the meetme.conf file as well as
Hi Sjaak,
> -Original Message-
> Arghh sure
>
> Something like
> Channel: ZAP/g1/xx/ZAP/g1/xxx/ZAP/g1/ etc
Ehm, not sure if that will work. But if you have an extension that does
Dial(Chan1&Chan2&..) you could use Channel: Local/[EMAIL PROTECTED]
> MaxRetries: 2
> RetryTim
You could do:
exten => 301,1,SetCallerID("Front Door" <301>)
exten => 301,2,
this should work.
On Thu, 6 Jan 2005 13:22:43 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:
> Hi list!
>
> I am going to install an intercom module for our home. The intercom can
> cater
All,
I have a machine with a TDM04B and a TDM13B card in it.
For over a week it was working find placing a call out every 10
minutes to the intercom system. Today that stopped working.
The call was still getting placed just as it was in the outgoing spool
and "looked" like everyting was working jus
Hi,
if not, what are my options (beside dedicating one line to capi4hylafax).
I'd like to cover faxes and voice on same channells - is this impossible
with capi ?
Thanks,
Rob.
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hello, using Asterisk, is there any clever way to
provide answer supervision based upon the received audio only from the FXO
interface (from a public PSTN switch that does not have battery reversal, or
CPC).
I would like to use something like:
[toCALLOUT;script to call a particular number
Is there a way to execute specific applications at Asterisk startup
(like "startup context" or something)?
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To UNSUBSCRIBE or update opt
Hi
I am using oh323-0.7.1 with asterisk cvs head version and works great
for me (Linux Fedora1), see details below:
Requirements:
PWLIB : pwlib-v1_6_6-src.tar.gz (or
Janus_Patch)
OpenH323: openh323-v1_13_5-src.tar.gz (or
Janus_Patch)
On Thu, 6 Jan 2005, pbx wrote:
> I just installed few TE410P in ml330/350 G3 dual Xeon 3.6ghz without
> any problem (kernel 2.6.8)
> Coulds you describe the problem more specificaly.
> Jack
Suggest you review the thread where you will find several explanations of
the problem. It's on G4 s
Steve,
It is my understanding that TxFAX does not allow to callout and send a
fax, but to be used in Faxpickup only.
Why would anyone implement something as dumb as that? I didn't :-)
Great :-)
Do you have it working in the manner you described, and if yes, could
you update the Wiki, or share you
On Thu, 6 Jan 2005, Steven Critchfield wrote:
> I don't remember for sure if you had tried it with IOAPIC turned off. I
> did look back and see that you only have 9 IRQs actually in use so the
> normal PIC should be fine. I would try that after a full power down and
> power disconnect. The power
They were updated, to reflect the new card. And I can call in perfectly.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [A
[EMAIL PROTECTED] is believed to have said:
>
>Hey Dan!!
>
>Give us a clue as to what hardware/setup & network provider you have there,
>and we might be able to help :)
>
>Paul
>
Hello Paul, hello everybody!
I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI
2.00, together
Rafael,
Thanks for the detailed instructions.
This really helps everyone looking fix this nagging issue.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rafael J.
Risco G.V.
Sent: Thursday, January 06, 2005 9:45 AM
To: Asterisk Users Mail
If this is a single CPU system try loading the smp kernel
"yum install kernel-smp" then force this to boot in grub.conf
This will force your interrupts to be handled with IO-APIC
handler instead of the old style XT-PIC interrupts handler.
Switching to APIC from XT will normally fix a lot of inter
FROM MY SIP.CONF
[1000]
type=friend
host=dynamic
context=local
allow=ulaw
secret=YESITIS
callerid="Front Desk" <1000>
[EMAIL PROTECTED]
dtmfmode=rfc2833
nat=0
FROM MY EXTENSION.CONF
[local]
include => mainmenu
include => parkedcalls
include => trunklocal
include => trunktollfree
include => tr
This is most likely due to the HUGE CHANGES in recent code in regards to
linked-lists. Be patient. They are being fixed and optimized.
-Matthew
- Original Message -
From: "rizwan" <[EMAIL PROTECTED]>
To:
Sent: Thursday, January 06, 2005 12:34 AM
Subject: [Asterisk-Users] asterisk addson
> You can run a simple mail server on the * box to accept emails addressed
> to the .fax domain (i.e. "[EMAIL PROTECTED]"). This presumes you are
Got documentation on setting this up?
> The second method is print-to-fax. This requires the configuration of a
> Samba "printer" on the * box.
Why can't people learn to turn off this crap for lists? I don't care if you
are on vacation and I'm sure that everyone else on this list doesn't care
either.
-Matthew
- Original Message -
From: "Dijkstra, Roelof" <[EMAIL PROTECTED]>
To: "Matthew Boehm" <[EMAIL PROTECTED]>
Sent: Thursday,
On Thu, 2005-01-06 at 09:19 -0600, Matthew Boehm wrote:
> > You can run a simple mail server on the * box to accept emails addressed
> > to the .fax domain (i.e. "[EMAIL PROTECTED]"). This presumes you are
>
> Got documentation on setting this up?
I'm getting round it at the moment using Hyl
On Thu, 6 Jan 2005, John Voss wrote:
> Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this
> posting.
>
> http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html
>
> I've tried copying the config in this listing with no success.
>
> One thing that I hav
its a GSM router. routes the calls from the GSM
network to a fxo device. can u pliz tell me what tx/rx
gains i should try out.
thanks
__
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.
http://promotions.yahoo.com/new_ma
its a GSM router. routes the calls from the GSM
network to a fxo device. can u pliz tell me what tx/rx
gains i should try out.
thanks
__
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
No it didn't.. it runs as root. REPEAT AFTER ME..
I WILL NOT INSTALL ASTERISK, LIBPRI or ZAPTEL from PORTAGE!
That's the only answers I'm a gentoo user too and I NEVER use portage
for those three things (well mpg123 also because 0.59r isn't in portage :P)
bkw
> -Original Message-
>
Hello, I have the TDM400P from Digium with an FXS on port 4 and FXO on
port 3. I am using the Xorcom asterisk debian self install. The FXS
works ok but the FXO comes back with ZT_CHANCONFIG failed on Channel 3:
Invalid Argument (22) - did you forget that FXS are configured as FXO
and FXO are config
Buddies works with up to 7 extensions it seems. Do a search, there
was a message just the other day about it.
On Thu, 06 Jan 2005 08:18:50 -0500, Joseph <[EMAIL PROTECTED]> wrote:
> Does anyone have the sip subscribe working with the IP600 phones?
>
> I saw a message about the Buddies, but I co
Lars Fredriksson wrote:
I have benn playing a little with quesues tonight and I found out if there
are at least one member-extension free the announcement with p'the place
in the queue wont be played to the person who called in.
This is a change that went into CVS (and changed the default behavior,
I'm having a problem with the call pickup with the latest CVS. Before I
updated to the latest CVS it was working fine. Now, whenever anyone
tries to pickup a call using *8 it dumps all calls going on at the time
and hangs up on the incoming call.
Tim
_
Noah Miller wrote:
I guess the phone just doesn't register as busy when there is only one
call on a line. It has to have two calls on a line appearance to
register as busy. Has anyone figured out how to disable this hold
feature and just have the second call go to the second line, the third
c
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kanuri,
If you want to use the last stable release of asterisk (1.0.3), you
should do it:
(don't forget to read the README )
Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz
Get pw
Hello everyone,
I have two internet connections available and I thought that it would
be really nice to be able to use both of them somehome (loadbalancing,
failover). I am familiar with lartc.org so I followed some of the
documentation here:
http://lartc.org/howto/lartc.rpdb.multiple-links.h
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 pho
i dont know how to use * native moh feature which is added recently to CVS HEAD
each time i hold a call i will get this warning on cli:
WARNING[24235]: res_musiconhold.c:837 local_ast_moh_start: No class: default
Paradise Dove
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Asterisk-Users mailing
Hello,
I am using * with latest bristuff. I tried to make use of ZapRAS
application (with the little or no documentation available), but I get
the following error when trying to establish RAS connection:
app_zapras.c:149 run_ras: wait4 returned -1: No child processes
Anyone has an idea what this
Hi all,
I'm trying to install a TDM400P card, and I need some
help.
Please, see below...
after dmesg command:
[EMAIL PROTECTED] root]# dmesgvia82cxxx: board #1 at
0xD800, IRQ 5Zapata Telephony Interface Registered on major 196PCI:
Found IRQ 3 for device 00:09.0PCI: Sharing IRQ 3 with 00
Tim,
For what it's worth, from my working sip.conf for Polycoms:
[2010]
type=friend
username=usr2010
callerid="MyName" <2010>
secret=nobodyknowswhatitis
host=dynamic
dtmfmode=inband
context=admin
defaultip=192.168.1.10
progressinband=no
Notes:
dtmfmode=inband and progressinband=no - that seems to b
I have following setup
Asterisk - T100P -> Adtran TSU600 P + FXOcard -> PSTN line
When PSTN line is plugged directly in to analog X100P caller id is
received by Asterisk
but when I plug it into adtran I'm not getting caller id.
Any ideas what kind of setup Adtran TSU600 requires to pass caller
Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.
Had this same behavior with IP500s last
week. For us, the solution was NAT related. Our server is on the LAN with
the phones, but I was doing 1-1 NAT of an IP on the outside to the server.
I changed t
Matthew Boehm wrote:
You can run a simple mail server on the * box to accept emails addressed
to the .fax domain (i.e. "[EMAIL PROTECTED]"). This presumes you are
Got documentation on setting this up?
Nope, I'm not in the documentation business. But you can go right down
to your local book
The 2100 has two ethernet ports
On Thu, 6 Jan 2005 09:29:48 +0100, Robert Rozman <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I've found approximate same pricing for both. Sipura 2100 seems to have more
> features...
>
> What are differences between those two ?What about their reliability
> (specia
Hi all.
Can anyone comment why shouldn't we use FC 3 for an * production system?
I'm not looking to start a distro war, but we just found out that redhat
9 (and FC 1) don't support SATA drives, and apparently FC 3 does.
We are only familiar with red hat and are in a point in time that
switching
John Middleton wrote:
Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk
from the digium web server - Whats the CVS command for a 'head'
install ?
Try "make datafiles"
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http
Has anybody looked into implementing a fax send interface for Asterisk using
the FSP code, that way it would plug straight into outlook and all the other
windows bits'n'pieces?
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It
Check your firmware on the Adtran 600.
It probably needs to be updated.
Tim.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Marcin izo
> Sent: Thursday, January 06, 2005 10:08 AM
> To: Asterisk-Users@lists.digium.com
> Subject: [A
This isn't a dialplan issue, it's a SIP issue. The same dialplan and
sip.conf are working perfectly with the other server.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 9:12 AM
To: Asterisk Users Mailing
Dan Adams wrote:
I was wondering, does anyone know if it is possible to have a stream of
audio coming from a Microsoft compressed audio stream fed to the caller
if they are placed on hold and if so how might this be done?
Write a shell script called mpg123, located in /usr/bin or
/usr/local/bin
Shoval Tomer wrote:
Hi all.
Can anyone comment why shouldn't we use FC 3 for an * production system?
I'm not looking to start a distro war, but we just found out that redhat
9 (and FC 1) don't support SATA drives, and apparently FC 3 does.
If you didn't have a problem with RH9 then why are you conc
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