Thank you all for the tips, I really appreciate it!
Now I need advice from someone running ISDN4Linux with Asterisk. ;)
Since I'll need to buy another card (US$45 including shipping is worth it),
which one is the best so I'll wont have any problems related to hardware
configuration? (remember,
hi everyone,
Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?
--
P. K.
[EMAIL PROTECTED]
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essentially, by not keeping to the same Call-ID or tag, asterisk has
no way of matching the wellgate's register with the past proxy auth
packet (and thus the hashed md5 token).
I've just gotten the box to register all 4-ports with an external SIP
provider. The provider is running an old
P. K. wrote:
hi everyone,
Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?
I've done just that - got it working with an X100 clone card, so I
figured it'll work just fine with the Digium cards too.
BTW I was also able to fax from
On Tue, 21 Sep 2004, P. K. wrote:
Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?
We have done it with the E1 PRI card in the Panaconic. The Asterisk box
sits between the pstn and the kx-td1232 with two E1 connections.
Peter
Peter Svensson wrote:
On Tue, 21 Sep 2004, P. K. wrote:
Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?
We have done it with the E1 PRI card in the Panaconic. The Asterisk box
sits between the pstn and the kx-td1232 with two E1
On Tue, 21 Sep 2004, el Flynn wrote:
PSTN - Panasonic - Asterisk
I realize that you'd have to do some programming on the TD1232 to
emulate DID or something like that, but it just goes to show how
flexible Asterisk is.
So you could hook up a 4-port FXO card from Digium, and program the
Hi,
I'm looking at connecting an analog fax to asterisk via an FXO card.
The plan is to send faxes thru freshtel.
Has anyone done faxing with freshtel?
Cheers,
-Shaun
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On 21/09/2004 15:07 Leo Ann Boon said the following:
I've just gotten the box to register all 4-ports with an external SIP
provider. The provider is running an old release of Broadsoft backend.
Seems like Broadsoft supports this strange way of authentication.
the 3504As do work with welltech's
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see
below lsmod),
does anybody has an idea what is wrong?
many thanks for help,
alex
snd:/usr/src/zaptelrtc# make
cc -c zaprtc.c -D__KERNEL__
At 10:05 21.09.2004, you wrote:
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see
below lsmod),
does anybody has an idea what is wrong?
many thanks for help,
alex
sorry, i found teh
Hello Leo,
3802 is doing the same (it is the fxo one)
--
Best regards,
Dannymailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet solutions
place princesse elisabeth 9/11 - 1030 Brussels - Belgium
Tel : +32-(0)2-215.67.65 -
On Tue, 2004-09-21 at 03:12, Atuc wrote:
At 10:05 21.09.2004, you wrote:
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see
below lsmod),
does anybody has an idea what is wrong?
Not sure about freshtel, but if it is a sip provider, make sure the
protocol is lossless, such as ulaw
Has anyone done faxing with freshtel?
Youness
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Hello
I am using a TDM400P-4FXO to connect my Asterisk to
telephone line. However, this TDM400P uses RJ45 connection while our telephone
standard uses RJ11. How can I wire the cable for the connection?
Thanks
Hoa
---
Outgoing mail is certified Virus Free.
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For
info
The
new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk
to make calls on the sip.btcommunicator.bt.net service. If anyone wants help
withthe settings, e-mail me off list.
:)
Peter
-Original Message-From: Whisker, Peter
[mailto:[EMAIL
On Tue, 2004-09-21 at 10:55 +0200, Nguyen Quang Hoa wrote:
Hello
I am using a TDM400P-4FXO to connect my Asterisk to telephone line.
However, this TDM400P uses RJ45 connection while our telephone
standard uses RJ11. How can I wire the cable for the connection?
The horrible answer is
Graham Turner [EMAIL PROTECTED] lazily top-posted:
i have installed asterisk / zaptel from cvs distribution as of 17/09/04
so i assume this does it
If you have a TDM/FXO then you'll need the latest CVS code. If you
have a X100P then you'll need any CVS (the latest is usually a good
choice)
Christian Victor schrieb:
I am trying to suppres the transmission of my CallerID when I place a
call using a .call file in /var/spool/asterisk/outgoing
Okay - now I have a little Progress. :-) Suppressing CallerID on a PRI
is done by setting the CallingPres parameter. But unfortunately this
I suppouse that you are using AgentLogin application, try instead to use
AgentCallbackLogin. Here is example how I do call center application on
my site:
[extension.conf]
---
[macro-agent-login]
; Agent login
; ${ARG1} - Caller nubmber - same as agent number
exten =
The standard way to get around this is to use the doublehash (or
maybe doublepound but unlikely to be doubleoctothorpe) patch which
will allow you to press hash twice for transfer or once to send it to
the remote end. IIRC you can also specify the timeout for it to wait
for the second
at last (using libtiff 3.5.7 as Mike Machado suggested) i was able to get
spandsp working on my debian sid 2.6.8.1, with * up to 18-08-2004 and
bristuff 4a
i've tested only fax receiving and the file on /var/spool/asterisk-fax/
contains only the first 3-4 cm. of page sent
sending fax notified
On 21/09/2004 16:13 Danny Zak said the following:
Hello Leo,
3802 is doing the same (it is the fxo one)
danny,
i've sent you the patch for 1.0-RC1 in a private email. could you apply
that, rebuild asterisk and the test it with the 3802 to see if the problem
goes away ?
--
Regards,
Hello,
i would like to use Meetme and i need zaptelrtc for that, since i dont
have any USB devices or a card from digium.
I compiled it on Llinux 2.6.8 and all i got was a zaprtc.o which obviously
wont work with a 2.6 kernel:
~/zaptelrtc# make load
sync
modprobe zaptel
insmod ./zaprtc.o
insmod:
Hello all,
I a'm having a lot of troubles compiling
the CAPI driver for mi AVM card, model C2 with two ports.
I´m using Debian stable with kernel
2.4.18 (bf24), but i can't compile this driver.
I just followed the steps from http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
and
why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it
in the zaptel make file and away you go =)
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Compiled Asterisk from FreeBSD port (0.9.0_2)
When I start asterisk it uses 100% cpu. Searches on Google
say to comment the noload = chan_oss.so in modules.conf
But this is already commented. Make.conf contains some
optimizations.
modules.conf:
; Asterisk configuration file
;
; Module Loader
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP
termination solution.
But who as it up and running ?
Best regards,
Han
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Hi all,
this is a rather newbie-oriented question, so please bear with me...
The system running Asterisk has been provided with an AVM FRITZ!Card
PnP. SuSE Linux 9.0 recognizes it right after booting the system and it
seems to be configured (MSN) correctly...
The hwinfo looks like this:
---
Hello all,
I a'm having a lot of troubles compiling the CAPI driver for mi AVM
card, model C2 with two ports.
I´m using Debian stable with kernel 2.4.18 (bf24), but i can't compile
this driver.
I just followed the steps from
Carefull plug in your RJ11 and it will work.
If you want to rewire to RJ45, use the middle two pins, 45.
Lyle
- Original Message -
From:
Nguyen Quang Hoa
To: [EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 3:55
AM
Subject: [Asterisk-Users] TDM400P: RJ45
why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it in
the zaptel make file and away you go =)
Zaptel Makefile:
MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \
ztdynamic ztd-eth wct1xxp wct4xxp ztdummy
~# lsmod
Module Size Used by
zaptel
On 21/09/2004 20:20 Jan Baggen said the following:
Compiled Asterisk from FreeBSD port (0.9.0_2)
When I start asterisk it uses 100% cpu. Searches on Google
say to comment the noload = chan_oss.so in modules.conf
But this is already commented. Make.conf contains some
optimizations.
add 'noload =
Hi All,
Here at my office we have two basic ISDN Access with two phone lines each.
We would like to install an Asterisk PBX. The question is: How can I connect the ISDN
lines to Asterisk?
Is it possible to connect to a DIGIUM Wildcard TE410P card? Or this E1 card is only
for PRI access? Any
Hello all,
My asterisk works well, the problem
is how to configure the AVM c2 card.
To provide ISDN external calls.
When you say, Just
make sure you have your kernel links set correctly,
what kind of links are you talking about?
Thanks.
Ismael Gil.
[EMAIL PROTECTED]
Enviado por:
I am taking a course in nework resource management with a theoritical
emphasis. Can someone suggest an asterisk related theoritcal project?
Thanks
M. Smadi
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On Tue, 2004-09-21 at 14:59, [EMAIL PROTECTED] wrote:
[snip]
~# lsmod
Module Size Used by
zaptel221764 0
crc_ccitt 2144 1 zaptel
md5 4000 1
ipv6 254980 12
fcpci 502616 0
capi
Hello all,
My asterisk works well, the problem is how to configure the AVM c2
card. To provide ISDN external calls.
When you say, Just make sure you have your kernel links set
correctly, what kind of links are you talking about?
The links in /usr/src/
Example:
lrwxrwxrwx 1 root src
I have some quetions/ideas for the Asterisk Call Queues system.
System information:
- Fedora Core 1
- Kernel 2.4.22-1.2115.nptl
- Asterisk CVS-HEAD-09/08/04-17:43:15
1. I sould like it that if a user is in the que and the expected wait
time is longer then xxx seconds or there are more then xxx
Hi Han,
Our company can offer you a SIP termination in Brazil up and
running.
Daniel
Johannes van Hulst wrote:
Is there an up and
running provider of SIP termination in Brazil?
I know that there are
some people building on a SIP
termination solution.
But
I sent this message last week, but it looks like it didn't go through.
So, if anyone receives this more than once, you have my sincere
apologies.
Hello,
I upgraded to CVS-HEAD-09/10/04-19:07:18 over the weekend and now
agents that are logged
in on zap channels have to acknowledge ACD calls by
Thanks,
My links, are properly set, as
you told me,
I just instaled Asterisk from de cvs,
first download it from the cvs and then compiling it.
Where could I find the asterisk header
files?
I just download asterisk in /usr/src/asterisk.
Regards from Madrid.
Ismael Gil.
[EMAIL
I dont know the card in question but
pri ISDN cards and BRI cards are two very different things
If you want to connect asterisk to an
ISDN 2e interface then you are going to need a BRI card (you can get dual
or quad interface cards)
You will also need to find out from
your telco how the ISDN 2
Hi,
I don't know much about the Wilcard TE410P but I surely know it won't work
with a BRI Interface ... as you suppose it's only for PRI access.
For a BRI Interface you can find several solutions, the cheapest would be to
simply use a BRI (ISDN) Card but there are also some more special cards
with
Not to flame a respond, but I only count 13 lines, not 200.
Anyway, what you posted is exactly what I am trying to prevent.
Do you see how you had to put 2 SetCIDNum entries for 2 seperate
dial-out numbers? Why can I not make 1 SetCIDNum entry for all
outgoing numbers below it like I tried to do
Hi. I'm getting new lines for using with Asterisk. In my Telco they said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences between RDSI and normal Analogic lines? Would I have
problems for using a
If someone takes a call from a queue on a CISCO 7960,
then does a Transfer to another agent (using the transfer button), the queue
system seems to think they still have the call, and wont assign them another
call till the other agent finishes the transferred call. Is this a known bug? Is
Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and running.
Daniel
IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio
de Janeiro.
Johannes van Hulst wrote:
Is there an up and running provider of SIP termination in Brazil?
I know
genius..pure genius..
ThomasNiesel=karma = karma++
Thanks,
Matthew
- Original Message -
From: Thomas Niesel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 12:48 AM
Subject: Re: [Asterisk-Users] ZapRTC
Olá Julio,
Também oferecemos IAX2.
Daniel
Julio Arruda wrote:
Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and
running.
Daniel
IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio
de Janeiro.
Johannes van Hulst wrote:
Is there an
I am interested too in termination using SIP to brazil, we need h.323 too...
Can you contact me?
Thanks
Sebastian.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Daniel Bichara
Enviado el: Martes, 21 de Septiembre de 2004 11:06 a.m.
Para: [EMAIL
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
-Original Message-
From: Christian Victor [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 4:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave [EMAIL PROTECTED] wrote:
Hi. I'm getting new lines for using with Asterisk. In my Telco they said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences
I am running a P4 2.8 with 1 gig of ram and 7200
rpm IDE drives. Nobottlenecks as yet.
Gary
- Original Message -
From:
Henry Devito
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Monday, September 20, 2004 8:25
PM
Subject: RE:
Hallo Martin Mielke
On Tue, 21 Sep 2004 14:32:34 +0200 you wrote:
...cut
so the isdn4linux drivers are correctly loaded. I know, CAPI should do
better but I can't compile from the tarball (see my post about it)
When trying to dial the PSTN using the ISDN interface I get:
---
*CLI
Hi all, i have installed two digium cards on my asterisk box a TDM04B TDM22B. The channels are configured as show below:
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Yes - that was what I thought too. But unfortunately leaving out the
parameter or setting it to '' will cause transmission of the default
number (usually subscriber number + 0)
On a PRI you
On Tue, 21 Sep 2004, Jay Milk wrote:
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I'm not the original poster, but I think this will not work. Just changing
the Calling Number (where the callerid field ends up in the isdn setup
On Tue, 2004-09-21 at 23:49, Matthew Boehm wrote:
Not to flame a respond, but I only count 13 lines, not 200.
That is 13 lines that *I* quoted from your email, you quoted a lot more
from the previous email (the entire email in fact)
Anyway, what you posted is exactly what I am trying to
After downloading the latest CVS head and testing it with the Cisco 7960
(SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid
audio dropouts.
I'm quite sure my gateway provider is running an older version of
Asterisk, and I suppose that this may be the root cause. But I
Most LEC's CLEC's, at least in our area, require sending a number (CSID)
before the call is completed. This is do to E911 features and ANI. If you
do not send a number the call will fail. If you truly want to block caller
ID I would contact your carrier and they should be able to block it on
Most LEC's CLEC's, at least in our area, require sending a number (CSID)
before the call is completed. This is do to E911 features and ANI. If you
do not send a number the call will fail. If you truly want to block caller
ID I would contact your carrier and they should be able to block it on
Matthew Boehm wrote:
Not to flame a respond, but I only count 13 lines, not 200.
It's still obnoxious.
All 200 of our extensions need to be seen to the outside world as the
same number (212-433-3344) but internally need to be seen as their
4 digit extension which has no outside world mapping
Hi!
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I'm not the original poster, but I think this will not work. Just changing
the Calling Number (where the callerid field ends up in the isdn setup
message) to nothing will most of the
Henry Devito schrieb:
Most LEC's CLEC's, at least in our area, require sending a number (CSID)
before the call is completed. This is do to E911 features and ANI. If you
do not send a number the call will fail. If you truly want to block caller
ID I would contact your carrier and they should be
Anybody ever managed to implement a solution where one could forward a
voicemail from one * server to another?
Dominique
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Thomas Niesel wrote:
[ snip ]
Does the phone had the same MSN?
I think so. It could dial outside without a problem...
Is there maybe a PBX needs a leading Digit to get outside line?
No, those are direct lines to the PSTN, so no leading 0 (or whatever)
is needed ...
Try your settings by
more detailed output
Sep 21 15:54:31 DEBUG[1120357296]: pbx.c:1255 pbx_extension_helper: Launching 'RxFAX'
Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1699 ast_set_read_format: Set channel
Zap/1-1 to read format SLINR
Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1666 ast_set_write_format: Set
Christian Victor wrote:
Wich is possible in Extensions by setting CallingPres=32. Now I am
looking for a way of disabling presentation in .call files.
Can you not just have your .call file dial back through an extension
passing in a parameter of the destination number, so that you can
activate
Unfortunately in my area call-by-call blocking caller id is not an option on
ISDN PRI, I did do a small work around for phones that I wanted to block
Caller ID from. I took one of my DID numbers and made the CSID that number,
I had the LEC not send a name with that number. I did not define that
Hello to all,
Im new user
of Asterisk. Im running Asterisk on a RedHat 9 platform. Everything seems
to be ok but I got lot of error messages and I dont know their meaning. Can
somebody help me ??
These are the
messages:
WARNING[163850]:
chan_sip.c:673 retrans_pkt: Maximum retries
Hi everyone,
Im having an odd problem with one of my sipura
boxes. The box registers the first time with asterisk properly after being
plugged in. After which, some of the subsequent registration tries fail and the
box becomes unregistered. However, after a few hours, it finally
HI all:
I have spent a large amount of time configuring/installing phones
connected to Asterisk. Halfway through the process I discovered that my
Cisco7960 with 2 7914 expansions was not supported in the SIP protocol.
After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of
Matthew Boehm [EMAIL PROTECTED] writes:
Do you see how you had to put 2 SetCIDNum entries for 2 seperate
dial-out numbers? Why can I not make 1 SetCIDNum entry for all
outgoing numbers below it like I tried to do with the 's' extension?
You can, you just did it the wrong way. ;-)
Is it
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a few minor hacks on
app_voicemail.c
-Original Message-
From: Dominique Kull
Hello,
We've released another update to our Asterisk GUI Client suite: 1.0.4
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
Why not just use rsync or netcat? There are about a dozen different
ways to do this.
John
Kris Boutilier wrote:
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery
Hi all,
I have a Wildcard that is flip floping between internally clocked and
the PRI. It is showing Red Alarm/Recovering. After a long run around
with the telco, they said I have lost the D channel on my side. I am
seeing this message:
== Restart on requested on entire span 1
Sep 21
I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if 'analogic'
means POTS then yes, he needs that ... TDM400P is an POTS/Analog NOT ISDN
device
--On Tuesday, September 21, 2004 11:28 -0300 Marconi Rivello
[EMAIL PROTECTED] wrote:
On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave
Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:
exten = 8899,1,Dial(SIP/8899,15,tr)
exten = 8899,2,Voicemail([EMAIL PROTECTED])
change it to
exten = 8899,1,Dial(SIP/8899,15,tr)
exten = 8899,2,Dial(IAX2/servername/extension)
We
I'm in the process of setting up a queue system where the position
message and thankyou message are required to play every 90 seconds.
However, if a caller comes in to a queue with active agents logged in,
and no one else is in the queue, the messages play immediately, and
then the agents are
Hello,
I am receiving an error in my error logs any time I receive a call on
the third line in our hunt group.
Sep 20 13:15:03 WARNING[1116939584]: Ring/Off-hook in strange state 6 on
channel 3
The weird part is that the calls seem to work fine, just this error
message is logged. Currently, I
If you are going to use the 7914 (which yes, unfortunatly isn't supported on
SIP, dammit Cisco) you might want to check out
http://chan-sccp.sourceforge.net
an alternative sccp module for *. Before we switched all our 7960's to SIP
we used this and it seemed alot better than the built in one.
This certainly works, if you want to have a remote VM - but still does
not forward a received VM to another server.
Dominique
Matthew Boehm wrote:
Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:
exten = 8899,1,Dial(SIP/8899,15,tr)
Agreed, however these rely on foreknowledge of the remote end configuration
and are non-transactional. I was thinking more along the lines of VPIM
(http://www.google.ca/search?q=%22Voice+Profile+for+Internet+Mail).
Consider a large-scale private networking scenario - it would be very nice
to have
Hallo Martin Mielke
On Tue, 21 Sep 2004 17:03:54 +0200 you wrote:
Thomas Niesel wrote:
[ snip ]
Does the phone had the same MSN?
I think so. It could dial outside without a problem...
Is there maybe a PBX needs a leading Digit to get outside line?
No, those are
Stable seized to exist quite some time ago.
On Tue, 14 Sep 2004 16:35:28 +0500, Atif Rasheed
[EMAIL PROTECTED] wrote:
on the asterisk site, it was stated while ago, how to download stable
version. like
cvs checkout -r v1-0_stable asterisk-addons zaptel libpri
but now it's not their. is
Hi everybody.
I have a Cisco 7905 IP Phone and as I see, the device isn't send the
registration message to the server, so to receive calls need to configure
static ip address.
Is there some way to make the Cisco send any sip registration? or Is there
some way to make the Cisco phone receive calls
hmmm I have no problems with 7960's and lates CVS since weeks
On Tue, 21 Sep 2004 10:41:54 -0400, Brian Cuthie [EMAIL PROTECTED] wrote:
After downloading the latest CVS head and testing it with the Cisco 7960
(SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid
audio
Hallo Daniel Eboa
On Tue, 21 Sep 2004 16:16:44 +0100 you wrote:
Hello to all,
I'm new user of Asterisk. I'm running Asterisk on a RedHat 9 platform.
Everything seems to be ok but I got lot of error messages and I don't
know their meaning. Can somebody help me ??
These are the
OK, that's it. I wont use the RDSI/ISDN connection and will get the
ANALOG :) (sorry about my english) lines.
Thanks a lot for your help.
RODOLFO
Michael Loftis wrote:
I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if
'analogic' means POTS then yes, he needs that ... TDM400P is an
Thanks Matt:
(damn cisco) :) == is right!!
I have already compiled the chan_sccp module. It is working just fine.
My main issue is actually configuring/loading the software the 7914 and
then using it like a main switchboard.
Thanks Again,
Jesse Tyler
On 21-Sep-04, at 9:53 AM, Matthew Boehm
Hi Jesse,
I would strongly recommend changing over to the SIP image and uisng
something like the Flash Operators Panel (www.asternic.org) instead of the
7914's. I experimented with chan_sccp2 a few weeks ago and decided that it
wasn't for me right now due to both the very limited support for the
Michael Bielicki wrote:
Stable seized to exist quite some time ago.
To expand on Michael's answer, stable wasn't being kept up to date like
it should have been, so the statement get the latest stable version
became get the latest cvs version as the standard answer for resolving
people's
Hey all,
Someone's posted one of my 800#'s on a poster in California for free
concert tickets, so I'm getting calls from California AC's at all times of
the day asking for tickets. I'm just using the 800# for friends and
family, and don't know anyone in these area codes, so I'd like to just
Nate Carlson wrote:
But if I try:
exten = 8005551212/408XXX,1,Congestion
exten = 8005551212/408XXX,2,Hangup()
It doesn't catch it. Is there any way to do something similar and allow
wildcards? Thanks!
See:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
On Tue, 21 Sep 2004 12:05:29 -0500 (CDT), Nate Carlson
[EMAIL PROTECTED] wrote:
Hey all,
Someone's posted one of my 800#'s on a poster in California for free
concert tickets, so I'm getting calls from California AC's at all times of
the day asking for tickets. I'm just using the 800# for
Hello All,
I am planning on setting up an * server for a customer and was hoping to get
a sanity check on my Plan. What I am trying to accomplish is a * voice and
16 data channel T-1 connection (ESF/B8ZS). I am planning on using a 2.8 ghz
P4, 1gig ram, on an Abit AS* Mobo, probably 3Com
I've had an IP300 for a while now and it's been working fine. I just
got an IP500 and when it connects to the FTP server it downloads the new
bootrom and says error loading.
The bootrom is fine and works on the 300... In addition, I downloaded a
new copy to be sure and it still doesn't work.
Hi,
I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver)
to work with an asterisk box.
The phone connects nicely to an external VoIP company (sipgate.de
reportedly using asterisk themselves) but there is a strange problem with
my asterisk:
- Incoming calls via ISDN
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