Thanks but I am aware of this method, I am trying to get the sequential
method to work.
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Kristian Kielhofner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent:
I have installed the CVS Head as of 12/12/04, as well
as the asterisk-addons to ensure that
/usr/lib/asterisk/modules/res_config_mysql.so exists.
I have configured the following (after building a new
DB with the appropriate SQL examples, with mods to
drop the invalid keys, on the Wiki):
-
Hi,
I wonder what does this warning 399 mean and how to workaround? sip show
peers says that sip client is unreachable althought it works with some
eexceptions ...
I saw posts in this list about setting codec to ilbc, is this right action ?
Also, I'm very interested if anyone succeded on
I have a problem with incoming
calls being recorded after a supervised transfer.
Incoming is CAPI BRI
- Asterisk - Supervised Transfer - SIP.
Call comes in, receptionist
answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold,
Callee picks up the call, Asterisk
I wonder what does this warning 399 mean and how to workaround? sip show
peers says that sip client is unreachable althought it works with some
eexceptions ...
http://www.google.com/search?q=%22detected+NAT+type+is+full+cone%22
___
Asterisk-Users
Anyone have an easy fix for making my music on hold to work properly?
It's very loud and has a lot of garbling in it. X is not running, and
the framebuffer is disabled.
I've tried just about every example I could find. I just uploaded
standard mp3's, but even the ones that came with it
It's very loud and has a lot of garbling in it.
What/how many phones have you tried it on? What channels
(ZAP/SIP/IAX2) and what codecs?
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To
Sorry... Im running SuSE 9.0
kido noagbodji wrote:
what os are you running?
K.
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion [EMAIL PROTECTED]
Sent: Monday, December 13, 2004 1:27 AM
Subject: Re:
Hi all,
is it possible to make a queue for outgoing calls? That's for preventing
Device '/dev/ttyI 0' is busy error when having only one line to dialout
and many files in /var/spool/asterisk/outgoing folder. So it would call
only one call at the time and when it's done it would move to next.
I get seg. fault with my * box. at the crash time i had about 35
Bridged Channel.
i have:
- dual xeon box (3.2Ghz)
- 2Gb of memory
- E7501 chipset motherboard.
- U320 scsi disks
- intel Gb ethernet device.
- i only use sip for clients (no fxs in box)
- TE405P for fxo (with 4 atran TA750).
- ulaw
Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad.
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
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To UNSUBSCRIBE or
Title: Message
Hi
all,
We are
located in Europe and we have four analog telephony lines.
What
hardware is needed to connect Asterisk with these lines?
WhatVoIP hard phones operate best with
Asterisk?
Regards,
Stojan
Sljivic
___
Asterisk-Users
A tdm40B 4 FXS card from digium. WE can deliver that to you, we are
even setting up a reseller in Belgrade. Please contact me off list for
details
On Mon, 13 Dec 2004 12:46:49 +0100, Stojan Sljivic - Pamet
[EMAIL PROTECTED] wrote:
Hi all,
We are located in Europe and we have four
We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p
EuroISDN.
We have 2 ISDN lines, one we had for testing, and one for general (40+
users) use.
During the testing phase, we had 10 phones linked to the second ISDN line,
and there were no problems with echo at all. Lucky
hi
is it, or can it be possible to transfer stuff like HANGUPCAUSE or
RDNIS over IAX2? This is really a nessicity for multi-server setups to
become any good...
roy
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[EMAIL PROTECTED]
Michael Bielicki wrote:
Stojan Sljivic wrote:
We are located in Europe and we have four analog telephony lines.
What hardware is needed to connect Asterisk with these lines?
A tdm40B 4 FXS card from digium. WE can deliver that to you, we are
even setting up a reseller in Belgrade.
Hopefully
On Mon, 13 Dec 2004, Roy Sigurd Karlsbakk wrote:
is it, or can it be possible to transfer stuff like HANGUPCAUSE or
RDNIS over IAX2? This is really a nessicity for multi-server setups to
become any good...
There is a patch floating around (on the mailing list and/or on the bug
tracker)
It looks like this is a splice between a couple of ISDN-30 lines and one or
more PBX's?
Are they both with the same provider, or with different providers?
We ended up adjusting the gain our ours as we would hear a distinct echo on
certain calls.
Other than that, you'll need to do the usual
Hi
Im trying to make Asterisk receiev SER calls and
then redirect them to GNUGK.
But until now, Asterisk isnt receiving
nothing...
Asterisk is already as a gateway in GNUGK as shown in the gnugk
monitorization:
is it, or can it be possible to transfer stuff like HANGUPCAUSE or
RDNIS over IAX2? This is really a nessicity for multi-server setups to
become any good...
There is a patch floating around (on the mailing list and/or on the bug
tracker) that transports the HANGUPCAUSE over IAX2 in a text message.
Hi,
-Original Message-
I get seg. fault with my * box. at the crash time i had about 35
Bridged Channel.
i have:
- dual xeon box (3.2Ghz)
- 2Gb of memory
- E7501 chipset motherboard.
- U320 scsi disks
- intel Gb ethernet device.
- i only use sip for clients (no fxs in box)
-
Are you replacing a Merlin Legend (hybrid PBX/key system) or a Merlin
4/10, 8/20 low end key system? You should be aware that in its current
form, Asterisk does not support shared extensions something commonly
used in most key environments.
/carmi
On Dec 6, 2004, at 9:37 AM, Pavel Jezek wrote:
Hi
When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all
But, when I register two or more clients to the SAME peer (with the same
user/pass) and I call to this peer.. Then only the UA which registered
the last will ring.. Others don't
On Mon, 13 Dec 2004 11:40:46 +, Andy Burns
[EMAIL PROTECTED] wrote:
If I have inbound SIP calls arriving from a provider's gateway to an
asterisk server on my LAN, which then routes the call back out via the
provider's gateway to a PSTN number, once the call is answered do all
the voice
Hopefully if you are setting up as a reseller you'll learn the diffence
between an FXS and an FXO, in this case a TDM04B not a TDM40B!
Because you expect resellers to know what they are talking about? That
would be nice, wouldn't it!
___
Asterisk-Users
Hi all:
I have install a E100P card. But when load the driver it reports error as
below:
[EMAIL PROTECTED] libpri]# modprobe wct1xxp
/lib/modules/2.4.18-3/misc/wct1xxp.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
greg
Regards
Greg Cirino
___
Cirelle Enterprises Inc.
603-425-2221
www.cirelle.com Web Application Development Design
www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster
On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote:
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
How about wctdm.c ?
--
Dave Cotton [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL
Sam Bashton wrote:
The data-heavy portion of the traffic is RTP, and that should be a
direct connection using your providers gateway.
Thanks, that was what I hoped for, no sense in all the traffic passing
up and down my ADSL to get back to where it came from, I suppose the
clue about SIP is in
Andrew Kohlsmith wrote:
On December 13, 2004 03:10 am, Soren Rathje wrote:
wait_just_a_bit(HZ/10);
I didn't want to wait inside the driver, likely a place where
interrupts are disabled...
Well, nobody claimed it was ready for production.. :-) I'm usually OK for
POC code, but don't
Just in case anybody missed it, the Broadvoice patch has been applied to
CVS HEAD:
=
Sat, 11 Dec 2004 23:33:48 -0600 (CST)
Modified Files:
chan_sip.c
Log Message:
Merge SIP authentication reuse patch (bug #2917) aka The Broadvoice
Patch with
Well, I know it isn't a classic style handset, but the Prestige 2000W is
small light and portable. If you have a suitable WiFi Card in your
laptop acting as an access point, you could squeeze a couple into
your laptop bag and do a pretty convincing demo without having to
hook into the client's
I'm using FC2. but with a fresh 2.6.9 kernel downloaded from kernel.org.
I've recently upgraded my Glibc to glibc-2.3.3-27.1.
I'm also using ECC Reg Memory.
and this is my Xeon CPU info: (HyperThreading is ON)
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model
Greg - Cirelle Enterprises wrote:
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
It was renamed to wctdm.c around Nov. 6. 2004
/Soren
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have found that because the way MyISAM works, InnoDB is a better solution to
prevent hangups due to the type of locking MyISAM uses. You have to edit
your /etc/mysql/my.cnf in order to enable InnoDB
From High Performance MySQL by O'Reilly :
innodb_data_file_path = ibdata1:400M
[EMAIL PROTECTED] wrote:
- dual xeon box (3.2Ghz)
- 2Gb of memory
- E7501 chipset motherboard.
- U320 scsi disks
- intel Gb ethernet device.
- i only use sip for clients (no fxs in box)
- TE405P for fxo (with 4 atran TA750).
- ulaw is used as codec and echo cancellationo is enabled.
Hi Guys,
I'm very interested if somebody using asterisk on
FreeBSD and not Linux without problem ?
Thank you for your feedbacks,
Ali
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To
If you do:
cvs checkout asterisk-addons
(without the -r v1-0), you'll get everything you need...including
res_mysql.conf.sample .
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 13 Dec 2004, Bill wrote:
Same here. I've deleted and re-installed
Have you examined your debug log for a possible SQL error? Updating user
passwords works fine on our systems. There should be no need to force anyone
to do it a certain way.
-Matthew
- Original Message -
From: Brad Hughes [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December
Both ISDN lines are going into the same * box - span 1 is the test isdn
line and span 3 is the live isdn line. The two ISDN lines are situated
right next to each other!
As mentioned there is no problem with the test line, so there isn't a
problem with * as such (I don't think!). Perhaps I haven't
Hello,
this is not possible,
you will have to solve this via the dialplan using parallel ringing or
queues.
Regards,
Marc
[EMAIL PROTECTED] wrote:
Hi
When I register a SIP or IAX client to asterisk and I dial to it from
another UA then there is no problem at all
But, when I register two or
It should be [settings] in your extconfig.conf not [default].
-Matthew
- Original Message -
From: Jason Goecke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 3:06 AM
Subject: [Asterisk-Users] Problems getting Asterisk Realtime to work
I have installed the CVS
More by trial and error, we backed off the gain until it disappeared but
with no detriment to the call quality (didn't want people to sound like a
whisper).
Our situation was somewhat different to yours though, we were seeing the
issues on calls from our PBX, not on calls through the IP phones.
Get newest CVS. Its in there. Trust me. Oh..be sure your getting
asterisk-addons.
-Matthew
- Original Message -
From: Greg - Cirelle Enterprises [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, December 12, 2004 12:50 PM
You are missing the fact that RealTime is not 1-0, its CVS. 'Thats' why
res_mysql.conf isn't even there.
-Matthew
- Original Message -
From: Bill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, December 13, 2004 8:32 AM
At 08:19 AM 12/13/04, you wrote:
On Mon, 2004-12-13 at 08:08 -0500, Greg - Cirelle Enterprises wrote:
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
How about wctdm.c ?
--
Dave Cotton [EMAIL PROTECTED]
Not sure what that is supposed to do but it
sure don't
On December 13, 2004 08:29 am, Soren Rathje wrote:
Well, nobody claimed it was ready for production.. :-) I'm usually OK for
POC code, but don't ask me to do production code, I haven't done any
serious coding the last ~25 years. I usually tell programmers what I want
and how I want it and
What's the proper way to download a STABLE version of asterisk and
asterisk-addons from CVS? I keep finding documentation that says two
different ways of download it.
Now that I've downloaded the asterisk-addons that has the
res_mysql.conf.sample it won't compile. If I cd to
Can somebody suggest theeasiestway to only allow outgoing long distance calls to countries x, y, and z?
Since Broadvoice allows free long distance to a bunch of countries I would like to take advantage of that, but block all other long distance calls.
Thanks,
Tom
Do you Yahoo!?
Yahoo! Mail -
[EMAIL PROTECTED] writes:
This is the last issue I have which makes that I can't get rid of the
SER proxy in front of asterisk.. Want to get rid of it
Out of curiosity, why?
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M
Because else people will complain that they can't register two
softphones anymore with same user/pass (because only one of the two
softphones can receive the incoming calls) :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Ivar
Helbekkmo
Sent:
Greg Boehnlein wrote:
On Thu, 9 Dec 2004, Jorge Mendoza wrote:
Andrei,
I'm interested too. Any chance to put the archive in a ftp site?.
Jorge Mendoza
I am also interested in getting the 1.3.4 firmware. It annoys me that I
can't just get it from Polycom's website, and forces me to rethink
Hello,
I would like to be able to dial in to my asterisk box.
Dial extension which would call two other people using the Sip channels.
We would like to be able to talk to each other at the same time.
Thanks
Bartosz Wegrzyn
___
Asterisk-Users mailing
Are there any others besides CVS and STABLE.
No. Only those 2. Unless I'm mistaken.
When someone downloads using cvs checkout -r v1-0 what version
is
that, CVS or stable?
The 1.0.* branch is refered to as STABLE. Anything above that, is called
CVS. (Also called CVS-HEAD)
Hi Jim,
Jim Van Meggelen wrote:
Getting dedicated IRQs for the cards is a minor problem compared to what
happens when you have four cards hammering away mercilessly at the
chipset and CPU of your motherboard; 1000 IRQs per second, per card.
Nobody's really sure what's wrong, but it causes
On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
I would like to be able to dial in to my asterisk box.
Dial extension which would call two other people using the Sip channels.
We would like to be able to talk to each other at the same time.
This is quite easy. :-)
Have the extension
I compiled version 1.0.3 over teh weekend on a Suse 9.1 box. It was a clean
installation straight out of the SUSE cds. Make sure that the kernel
sources are loaded and that you do a full online update before you proceed.
Asterisk compiles without any problems.
Vassilis
At 17:44 13/12/2004, you
On 13 Dec 2004 at 12:44, Rick Green wrote:
I am trying to do my first asterisk install on a SuSE 9.1 box, using
the asterisk-update script mentioned a few days ago on this list.
I did read the 'quickstart' document on onlamp.com, and made sure
the
following packages were installed
On December 13, 2004 11:53 am, Stephen R. Besch wrote:
1) This is not to minimize the problem, but 1000 interrupts per second
is quite a few, but not an overwhelming amount. Keep in mind that an
unbuffered serial card (and there are more than a few of these out
there) working at 19.2 Kbaud
Hello all,
So i am new to asterisk and very green when it comes to Linux, so don't beat on
me too bad :)
I just set up * on Red Hat 9.0 last night... everything seems to be
configured coffectly, I can start * no problem and get the CLI prompt... now
here is my question... I have an account
Well.. subject says it all really.
I have a TDM with 2 FXS modules and 1 FXO and a X100P.
If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine
but not the X100P
I have tried several combinations of port numbering but can some kind
person with a similar setup to send me the
I'm very interested if somebody using asterisk on
FreeBSD and not Linux without problem ?
I think a lot depends on whether you need hardware interfaces or not.
For voIP only I had no problem on FreebSD 4.5
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Asterisk-Users mailing list
[EMAIL
Hello all,
Is it possible to send dtmf tones to an answering terminal (after answering
the call)?
I have for example a external voicemail system that I want to connect to *.
Now for the right integration I need to send dtmf tones to the analog ports
that answered the call.
Cheers.
Robin
I have been looking to see if this type of phone can be implimented in
*. I have found nothing conclusive. Is any out there using a
multiline / mutlifunction phone typically used by a receptionist for
transfering / routing calls? I need to know how this is accomplished
or what alternative
I am having a similar problem for my home setup. I receive a call from
the PSTN and have * automatically dial one or both Cisco 7940's running
SIP firmware. In my case the callerid info just says asterisk as the
PBX is actually placing the call on behalf of the incoming PSTN call on an
X100P
I can't speak for the 7912G, but I have several 7905G phones and these
work perfectly with Asterisk.
This is great! The 7905G is what I have in mind for a plain basic phone
and the 7940 where a speaker phone is needed.
The firmware is easy to obtain if you have a Cisco support agreement -
I have Asterisk talking to MySQL using Realtime but for some reason I
keep getting the wrong context used when Realtime makes the MySQL call. I
can see this in my /var/log/mysql.log file. Because of this I can't login to
VoicemailMain from my X-Ten phone. I can login if I statically configure
It would be nice for Mark to comment on this design flaw ...?
Why so quick to assume it's a flaw? Perhaps it's a compromise.
Greg
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To UNSUBSCRIBE or
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Anyone know of a service provider offering this?
-Mark
707-735-1038
___
Asterisk-Users mailing
Mark Halverson wrote:
Sorry if this is the wrong list...
I need a toll-free number to be delivered to me on IAX. (This is NOT an
existing number need to buy the whole service.)
Anyone know of a service provider offering this?
-Mark
707-735-1038
ask this on the -biz list.
-Chris
--
Christopher L.
A..now we find the problem.
Voicemailmain does NOT use the context that calls it.
haha..finally found the problem.
You must call it as VoicemailMain(@from-sip) if you want it to look for
mailbox in a specific context.
I knew it was something simple.
-Matthew
- Original
In my queue I have about 4 agents answering at any given time, * has a
tendency of rininging the first agent (rrmemory) for only half a ring
then moving to the next agent, on the console it says it tried them for
20seconds. Anyone seen this or know where to look to fix it?
Thanks,
-Ryan
On Mon, 2004-12-13 at 18:47 +, Vassilis Konstantinou wrote:
Well.. subject says it all really.
I have a TDM with 2 FXS modules and 1 FXO and a X100P.
If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine
but not the X100P
I have tried several combinations of port
I'm very interested if somebody using asterisk on FreeBSD and not Linux
without problem ?
many of us are using * on 5.3-stable and 6.0-current. without a
problem would be a bit pollyanna-like.
randy
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Check out RealTime. This is how its done.
Otherwise..not really..each module has its own memory space and runs in its
own thread and so you can't share resources like that across memory space.
What are you using to query?
-Matthew
- Original Message -
From: Roy Sigurd Karlsbakk [EMAIL
At 09:32 AM 12/13/04, you wrote:
Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never works. The best I've gotten is the MySQL query to
execute with the wrong context. When I use cvs checkout -r v1-0 zaptel
libpri asterisk asterisk-addons
Yep, same problem I had. Look in /etc/mysql/my.cnf for the location of the
sock file.
On Monday 13 December 2004 03:17 pm, Greg - Cirelle Enterprises wrote:
At 09:32 AM 12/13/04, you wrote:
Same here. I've deleted and re-installed asterisk a few times and the
RealTime voicemail never
Getting dedicated IRQs for the cards is a minor problem compared to what
happens when you have four cards hammering away mercilessly at the
chipset and CPU of your motherboard; 1000 IRQs per second, per card.
Nobody's really sure what's wrong, but it causes problems for pretty
nearly
Even though you can...why would you? You can't use some things that are in
CVS addons with STABLE asterisk.
res_config_mysql.c and res_mysql.conf are part of the CVS version of
asterisk. This means that you cannot use them with STABLE.
If you want RealTime functionality you HAVE to upgrade your
Is there anyway to determine if is line is already in use by
another device such as a fax machine if the fax machine is not tunneled through
asterisk via a FXS out on a FXO? Right now * tries to pickup the line and dial
when I try the Channel Available command on the Zap FXO. Also, is
http://www.millenigence.com/articles/asterisk-non-technical-review.pdf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sean Cook
Sent: Monday, December 13, 2004 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Sean Cook wrote:
The company I work for is looking at vendors for a PBX, one of the
requirements is VoIP. I have been sitting there listening to people
pitch very proprietary implementations of VoIP where you are locked in
to their hardware, their interface...
I know a little bit about asterisk
On Mon, Dec 13, 2004 at 03:15:18PM -0600, Grady Trew, Jr. wrote:
Getting dedicated IRQs for the cards is a minor problem compared to what
happens when you have four cards hammering away mercilessly at the
chipset and CPU of your motherboard; 1000 IRQs per second, per card.
Nobody's really
I downloaded phpconfig and set it up to read my config files, but it never
successfully recognizes any of my sections. The regular expression seems to
be included in the line:
if(preg_match(/^\s*\[([^\]]*)\].*[\r\n]\$/, $line))
and later, the same regex.
I'm not sure about the [\r\n] on the end
I need to reopen this discussion because it's impossible to run spandsp
(and VoIP) under these circumstances.
With zaptel unloaded, I see the following vmstat 1 output:
no swapping, an occasional disk output, +/- 1003 interrupts/sec., less
than 10 context switches/sec., CPU idle 100%. A very
SER I am not happy with it.. didn't manage to get these below things
functioning:
It's too strict with authentication (user has to set specific
domain/realm) have problems with several types of hardphones authing to
SER
You can't make config changes without having to restart SER
Can't
Check your mpg123 make sure its not just a Symlink to mpg321
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Austad
Sent: Monday, December 13, 2004 9:55 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re:
http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Digium T100P, T1 cable to Adtran T1 port, extensions to Adtran FXS
interfaces. Follow the instructions on the Wiki for configuring the
T100P both in /etc/asterisk/zapata.conf and /etc/zaptel.conf. Configure
the ports on the Adtran per the
You can use a T1 from digium for that
(http://www.digium.com/index.php?menu=wildcard_t100p).
Please post using plain text next time.
From: Robert Augustyn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 14, 2004 12:03 AM
To: Asterisk Users Mailing
Hi,
I don't think any SIP server would allow you to register more than once with
the same login information. What you can do in asterisk is setup two
different entries in sip.conf and then use extensions.conf to dial both.
Example from extensions.conf
[default]
exten =
At 04:59 PM 12/13/04, you wrote:
Can anyone give me some recommendations for IP phones that work well with
Asterisk?
I'm hoping for something not much more then $100 bux or so.
grandstream bt100 will work 100
Also does vonage service work directly through Asterisk or would I have to
use their
OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0
asterisk' into 2 seperate directories.
I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source
code line differences between the two.
Some code that was in asterisk-cvs wasn't in asterisk-1.0.3 and vice versa.
Dear group members,
Somewhere in this representation:
http://graphics.cs.uni-sb.de/VCORE/Publications/mark_spencer/mark.smil it is
mentioned that one can cal an Mp3 file. How is this implemented? When this
Mp3 is playing, is it then still possible to receive a call?
Thanks,
Willy
-Original
Matthew Boehm wrote:
OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0
asterisk' into 2 seperate directories.
I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source
code line differences between the two.
Some code that was in asterisk-cvs wasn't in
Hi,
12.2. most have bugs. You need to check version. Also you may want
to try setting up a second voice codec and add alaw / ulaw as your first
preferences. This may work? But I think your biggest problem is your ios
version.
Ps
Don't forget to add you new voice codec preferences under
On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly:
I have been looking to see if this type of phone can be implimented in
*. I have found nothing conclusive. Is any out there using a multiline
/ mutlifunction phone typically used by a receptionist for transfering /
Has anyone been able to make the multitech voip box speak H323 with
asterisk? I am using the asterisk CVS from a week ago and the recommended
versions of pwlib and openh323. I am able to connect to the multitech 800
box at our remote office which is connected via POTS to a proprietary PBX
system.
Anyone help me here? I am a newbie so be gentle
;-)..
It worked once and then I played with the
configs.
I have a static IP address which is on my private
network.. Phone is 192.192.192.220 and asterisk server is
192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest
is )
When somebody call me on my pstn # cable connected to my fxo
card it does not work when I check my computer the following error shows
Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently
running on asterisk1 (pid = 2160)
Verbosity is atleast 3
-- Remote UNIX connection
--
Anyone using IAX.cc / Sixtel? Would love to hear experiences
good or bad.
Aren't you the competition? ;-)
Either way, I'm using a DID from them and have had no problems with inbound
calls, works a treat :-)
Cheers
Paul
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