In article [EMAIL PROTECTED],
taf taffey [EMAIL PROTECTED] wrote:
Done!
Going back to the issue at hand I've read the reply
from Tony in a previous mail and this is related to
call files. Is there a way to dial two
outbound/external numbers and bridge them together
using the Asterisk API
That would be nice, SER does have the possibility to answer an OPTIONS
correctly, but * indeed answers with a 404, i'm now using sipsak to
register a test user, that also works.
Andres wrote:
Erik Versaevel wrote:
Hello all,
Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check
I'm considering put this on the voip-info.org Wiki, but I thought I'd
throw it out a few observations here first:
* Cisco IP Phones are designed for enterprise deployments.
They are designed to be provisioned by the hundred or thousand. They are
not designed to be deployed for a single user or
Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router
using MGCP IP protocol, instead of controlling it through an E1.
Have anyone tried this configuration? How does MGCP works? I've tried to search
for it on Google, but I only find the protocol specification for it.
Is
John
We have no experience of providing a tool like the one you describe, but I'd
guess that AGI using sox/soxmix to edit recorded files should be able to get
you where you want.
The Trim options of soxmix provides the ability to, for example, take the
first x.y seconds of a recorded file so
What needs to be done to make this work?
For me, this would be the only time we'd really use attended transfers,
on the way from an agent to either another agent, or a member of staff.
At the moment we have to make all transfers from agents (i.e. queue
calls) via blind transfer.
Ben
Good day all
I have a asterisk server running sip and sip phone
How do I get asterisk to call another h323 server?
Please Help
Thanks
Altus
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Hi,
-Original Message-
Short of buying a (no doubt) expensive one designed
specifically for the
Cisco 7960, what are my options for using headset with this phone? Is
there some kind of adapter to buy so I can use standard
Plantronics/Jabra headsets? Is there by any chance a
Good day all
Just to re phrase my previous question
We have asterisk running sip for sip phone
In the US there is a h323 server
What I want to do is:
All calls coming into my pbx via sip thats got a american number to go
threw the h323 server
I have set this up with 2 sip servers where the one
Will be good, if somebody could provide rpms for every release and
also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules
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Did you already tried if the latest firmware version 3.56m for the snom190
solves the problem?
Regards
Nils Ohlmeier
On Friday 21 January 2005 02:35, Michael Di Martino wrote:
I have the dtmfmode in sip.conf set to use rfc 2833
however, when my users have to enter pin numbers to join let
On Fri, 2005-01-21 at 09:06 +0100, Daniel Nyström wrote:
Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP
router using MGCP IP protocol, instead of controlling it through an
E1.
Have anyone tried this configuration? How does MGCP works? I've tried
to search for it on
On Friday 21 January 2005 06:11, Andres wrote:
Erik Versaevel wrote:
Hello all,
Can asterisk itself respond to a SIP OPTIONS packet? i wan't to check
if asterisk is still alive by using sipsak (because of nagois mon)
Sure it does. It answers with 404 Not Found. We monitor all our SER
Hi Glenn,
What do you mean by provisioning?
Thanks
Mike
On Fri, 21 Jan 2005 03:02:45 -0500, Glenn Powers [EMAIL PROTECTED] wrote:
provision a hundred, so the process for provisioning one is going to
seem a bit overwhelming.
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hello
Sorry for my poor english...
A have probleme with callerid with isdn4linux
I am using diva pci 2.02 passive card,
Asterisk 1.0.4 (i have tested all releases)
Fedora core 2
Zaptel card
All work great but i have not the caller id.
I have trace chan_modem_i4l.c, and callerid id string send
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
i have encountered RESTARTs during call establishing which occurs
under irregular conditions:
The behaviour is like described below:
When an outgoing call is dialed a SETUP isdn message is send to the
telco's switch. The teleco's switch
answers
Matt Riddell wrote:
The easiest way to see that changes is to download one of the packages
from the Digium FTP site and read the CHANGELOG.
Or carry on reading!
ChangeLog:
Asterisk 1.0.4
-- general
-- fix memory leak evident with extensive use of variables
-- update IAXy firmware to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I'm having some problems using the asterisk modem driver with an
ISDN4Linux card (AVM Fritz). (I realise that it's better to use CAPI,
but unfortunately this card doesn't have any CAPI drivers).
Every so often the ISDN just stops working (it
Title: Message
Hi
all,
We have the
full logging enabled on one of our client's servers running stable 1.0.3 with a TE410P and T1
trunks usingWink Start, 99% of the
calls being inbound. I've noticed in the logs that the following three DEBUG
messages get repeated quite often, around 2000
Hi all,
Hopefully someone can help me.
I have some free 3Com SIP Phones and am trying to get working with *.
When I try to register the phone it gets a 403 Forbidden back.
Can anyone tell me if they have had success with these phones.
The firmware version is 1.0.5.0 but since the phones are
Hello all,
I'm a newbie in * and i want to start by making
internall calls between ip phones (Grandstream BT100,
and HT286),
if someone can help me with an ewample of sip.conf
file specially with the register field in [general]
defintion.
Thanks
Découvrez
Title: Message
Hi,
Ihave stress tested the Asterisk Voicemail.
We
have encountered problem with simultaneous calls that are sent to the same
mailbox.
It
occurred that several calls were writing to the same file.
It
seems that there is a synchronization issue in the Voicemail
application.
In short,
Switching traffic between wholesale providers (my customers) is what I
want, collect CDRs and bill them =)
I have no gateways of my own so Im not originating nor terminating calls,
just switching traffic is my goal, all this people use h.323 of course.
Thanks for your help
D.
On Fri, 21 Jan 2005, Daniel Nyström wrote:
Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router
using MGCP IP protocol, instead of controlling it through an E1.
Have anyone tried this configuration? How does MGCP works? I've tried to
search for it on Google, but I only
Is it possible to record a meetme conference? What
channel would you monitor, is there a main channel that all audio goes too?
If so, is it possible to use the ast_monitor (iirc)
to record that channel?
Cheers,
Ben
___
AFAIK, Cisco 79xx phones don't have web-based configuration. They have a
telnet interface, though it's enabled/disabled based on the config files the
phone
gets from the TFTP server.
FWIW, my 7905A with SIP firmware 1.01.00(030807A) has web but not
telnet. The web interface has a number of
Good day all
I cant get my grandstream bt-100 to register
My asterisk is on a public ip and the phone behind a nat firewall
I added nat=yes in sip.conf and did this on my grandstream
set the GS to SIP server=asterisk.yourhost.com and leave Outbound
Proxy empty
* set the GS to SIP port 5060 and
The thing about PCI-cards were just a way to illustrate how I'm thinking, not a
sollution. I'm sorry for the confusion i brought.
It's already clear that an Adit 600 will work as an Channel Bank.
The problem was just that, either we use on E1 to the Adit, but there might be
some issues with
Has anyone used one of these with *, any observations/comments please?
Thanks
John
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Hey I am intrested in testing this thing out , i was already trying to
work out something of my own when i found this ,s o can we work on
this?
~uppal
On Wed, 19 Jan 2005 11:23:26 +0200, Mike Tkachuk [EMAIL PROTECTED] wrote:
Hello All,
Want to say that it not latest version of this tool.
Title: Message
I tested
this as well and found the same thing.
I
have submitted a bug report:
http://bugs.digium.com/bug_view_page.php?bug_id=0003394
That
one's severe enough that I have flagged it as a major. On a busy system, this
bug could cause all kinds of messages to go missing,
We record meetme rooms by sending a manager Action to place a call from the
meetme room to an extension that is defined to start recording for a
predetermined amount of time, to end that recording we just send an Action
to Hangup that channel. Been working great for over a year now with over
Hi,
asterisk does not send the RESTART message, the switch sends the message
right after the SETUP ACKNOWLEDGE message, note the s in the pri
debug output.
regards
Klaus
Am Freitag, den 21.01.2005, 11:09 +0100 schrieb hanson:
When an outgoing call is dialed a SETUP isdn message is send to
Hello,
I've tried all the Wiki pages and still can't seem to
get this thing working and that's why I've posted this
mail.
I would like to dial two external numbers and
conference them together using the asterisk api
manager.
Hint: search the wiki for local channels
--
Nicolás Gudiño
Am Freitag, den 21.01.2005, 13:48 +0100 schrieb Hannes Kepplinger:
Klaus,
thanks for your quick reply. I thought that Originator or Terminator
shows the direction.
Danke, Hannes
Hannes,
yes it does, but you also have to look at the call reference. Basically
the RESTART message from
Do you think it's hearable? All communication will be on a dedicated Fast
Ethernet link (just a cross-over cable). And it will still use aLaw codec (same
as Euro ISDN afaik).
Since E1 = 2048Kbps and FastEth = 100Mbps, I concidered it fast enough to not
make any latency. It seems like MGCP also
hi,
is there a possibility to provide german dialtones on an IAXy S100IPWRD?
'language=de' sets only the messages to german (voicemail, etc.)
is there something like 'loadzone' as in /etc/zaptel.conf
regards
frank
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First things first -- don't reply to a message about something COMPLETELY
different, erase everything and start your new message. Just click on the
To and start your new message.
When you reply and erase everything you are unintentionally placing your
message in the middle of an existing
I made one sever with the XORCOM RAPID
will you tell me how can i use it with softphones? and
from where i can download that
actually am a newbie on this. so please give me help
this is to work on a lan. so how can i do that
hope that you can give me a clear help or clear link
with regards
I'm having the exact same issue on a brand new Dell Poweredge 700, using
FC2. It locks the machine totally.
-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Thanks to Bruce for adding this stuff on attended transfers to the WIKI
pages. I've been trying to get my head round this for a couple of days.
Unfortunately I'm still having a bit of trouble.
I have the latest CVS-HEAD, just downloaded and compiled. Added the bit for
attended transfer into
Please, development is important (of course) but updated documentation
is important too!
is there any special guide followed by * developers about how to
document funcionality? I can take care (and will do so happily) of
trying to update the documentation at asterskdocs but i must obtain
the how
As far as I understand, if you completely doing VoIP without any PSTN
intervention, in this case
it's probably unregulated. I case of PSTN-to-VoIP gateway - this is
completely different story.
Here, in Canada, you have to have International Basic License Class A,
to provide (excerpt from
On Fri, 21 Jan 2005, Daniel Nyström wrote:
Do you think it's hearable? All communication will be on a dedicated
Fast Ethernet link (just a cross-over cable). And it will still use aLaw
codec (same as Euro ISDN afaik).
Since E1 = 2048Kbps and FastEth = 100Mbps, I concidered it fast enough
to
Is the broadvoice patch part of asterisk 1.0.4?
The changelog does not mention it.
Thanks,
Jerry
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- Original Message -
From: Jean-Michel Hiver
If I understand, you are doing.
Extension (VoIP) - (VoIP) Asterisk (VoIP) - (VoIP) Zyxtel (FXS) -
(FXS) GSM
Unless I've missed something, it seems that you are trying to plug a
telephone adapter (Zyxtel) to another telephone adapter
Where can i get information about Asterisk using Radius?
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Rich Adamson wrote:
I'm not a SER user, therefore others on this list might have a better
understanding as to its appropriateness.
Other possible approaches:
- two * systems, one of which is colocated outside your corp structure
with iax link, and a sip client with two proxy registration
I'm new to this as well. I use SJPhone as my software. Works good. The
default install of Asterisk has allot extensions and voice mail boxes to
play with.
http://google.com asterisk sjphone and you will find allot of good
stuff.
David
On Fri, 2005-01-21 at 05:20, Krishnan wrote:
I made one
On Fri, Jan 21, 2005 at 12:24:32AM +0100, Marco Vescovi wrote:
reading around and surfing the net I've found some informations about QSIG
PRI protocol, that seems a good choice to integrate 2 PBX systems with PRI
interfaces. The question is: which is the state of Asterisk support for that
Ups!
I was not aware of it, thank you for pointing it out
and thanks for the info.
robert
--- Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
First things first -- don't reply to a message about
something COMPLETELY
different, erase everything and start your new
message. Just click on the
To
I second that, we should write to Dag and Thias, they compile so much stuff.
On Fri, 21 Jan 2005 13:23:37 +0400, VX Lists [EMAIL PROTECTED] wrote:
Will be good, if somebody could provide rpms for every release and
also rpm's with static compiled chan_oh323 and Asterisk-oh323 modules
Hello, I have two TE-405Ps that I am having trouble with.
I'm using an Intel 865 motherboard with a Celeron D processor. Kernel 2.4.26,
Slackware 10.0.
my /proc/interrupts:
CPU0
0: 172317 XT-PIC timer
1: 2 XT-PIC keyboard
2: 0
Try the Wiki, thats where I found it.
John Dunham
Original Message
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com,
Subject: RE: [Asterisk-Users] Asterisk + Radius
Date: Fri, 21 Jan 2005 10:24:20 -0500 (EST)
Where can i get information about Asterisk using Radius?
Peter Svensson wrote:
On Thu, 20 Jan 2005, Steve Clark wrote:
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
snip
Thanks Peter,
No offense to the person who maintains the changelog, but it rarely contains
the information I want to know. This is what I use to find out the changes:
cd /tmp/
cvs co -r v1-0-4 asterisk
diff -ur /usr/src/asterisk /tmp/asterisk
This shows code level differences between what I am using in prod
www.voip-info.org
On Fri, 21 Jan 2005 11:30:51 +0100 (CET), ihsane moutaib
[EMAIL PROTECTED] wrote:
Hello all,
I'm a newbie in * and i want to start by making
internall calls between ip phones (Grandstream BT100,
and HT286),
if someone can help me with an ewample of sip.conf
file
Since the provisioning can be done on the phone itself, I think what
you are writing is not true. If one is using it in a SOHO environment,
one can just provision it from the phone.
On Fri, 21 Jan 2005 03:02:45 -0500, Glenn Powers [EMAIL PROTECTED] wrote:
I'm considering put this on the
Christopher wrote:
Thanks guys, really appreciate the responses.
Actually I've tried the suggestions in this document with absolutely
no luck at all unfortunately, and turning off fixup protocol udp sip
was the key to allowing my remote phone to ring to an internal phone
(when fixup is on I
As CarrierAccess states, there can be potential mismatch regarding the TDM
signaling required to terminate the voice channels onto the FXS cards.
I'm not sure I understand this fully.
He also says Although E1 still remains as another option, given a compatible
signaling pattern. Since I'm a real
I just installed 1.0.4. When I do a show version, it still says:
Asterisk CVS-v1-0-12/21/04-14:14:46 built by [EMAIL PROTECTED] on a i686
running Linux
Which is exactly what is said before the upgrade. Is this right?
-Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Henry Devito wrote:
www.thirdlane.com http://www.thirdlane.com has already written a
close dsource webmin module. I have no idea how much it costs or how
well it works.
I've attempted to contact thirdlane to get pricing on their GUI and
can't seem to get anyone to reply.
My personal
On January 21, 2005 10:13 am, [EMAIL PROTECTED] wrote:
Hello, I have two TE-405Ps that I am having trouble with.
I'm using an Intel 865 motherboard with a Celeron D processor. Kernel
2.4.26, Slackware 10.0.
9: 0 XT-PIC t4xxp
That is a bad thing... not only are you on
http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
a mailing list looks like to most people, and you can see why
replying to a message, erasing its contents and starting an
entirely new email about a different topic is frowned upon
(yours is the highlighted message).
I know this
I have not used the adit 600 with asterisk, but have used it with a voip
softswitch trunking via mgcp. i currently have 5 fxs cards with 8 ports per
card in an adit unit with a cmg card. i think the price was like $3000 for
the unit. it works great but im about to take it out of service and
You can even make your own adapter if it has to be really cheap :)
I saw that on the Wiki a few moments after I posted the initial query,
but I had a question: do you know how I could make that adapter if I
wanted to use a single 2.5mm connector headset (like the kind used with
cellphones and
Any T extensions set?
Maybe autofallthrough=yes and absolutetimeout
On Fri, 21 Jan 2005 17:02:44 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
I have a zap line on a X101P which will occasionally just hang up the
call for no apparent reason. Is there any good way of trying to
diagnose what
Hi,
I have a problem calling with SIP Communicator.
Connecting directly from SIP Communicator to the VOIP Provider, the
audio doesn't work.
Connecting throught Asterisk, the audio doesn't work.
Connecting throught Brekeke, all works correctly.
I need to use Asterisk and SIP Communicator. Does
If you look hard enough in the outlook options, you would see that there
is a threaded view. Look harder!
On Fri, 2005-01-21 at 10:38 -0500, Nabeel Jafferali wrote:
http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
a mailing list looks like to most people, and you can see why
On January 21, 2005 10:38 am, Nabeel Jafferali wrote:
http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
a mailing list looks like to most people, and you can see why
replying to a message, erasing its contents and starting an
entirely new email about a different topic is
Ok,
Answering my own question. The problem is that the sipura does not support
some of the sip compact header format that is now used by CVS-HEAD. I have
been sent a quick fix version of the firmware but it will be generally
available in the next firmware release.
Chris
- Original Message
I'm playing with [EMAIL PROTECTED] and it has a web interface. Great for
users to check Voicemail. Anyways I would help with webmin but I have
never wrote html before and I'm very new to Asterisk.
Thanks, David
On Fri, 2005-01-21 at 07:27, [EMAIL PROTECTED] wrote:
Henry Devito wrote:
I'm running Asterisk CVS-v1-0-12/20/04.
I'm using PHP with Manager API Here is the code:
# Make call
$socket = fsockopen($ask_db,5038, $errno, $errstr,
On Fri, 21 Jan 2005, Daniel Nyström wrote:
As CarrierAccess states, there can be potential mismatch regarding the
TDM signaling required to terminate the voice channels onto the FXS
cards.
I'm not sure I understand this fully.
You can run a number of signalling protocols over a channelized
morn all,
I am trying to compile 1.0.4 and SpanDSPpre10 to compile and have problems
patching the apps makefile. Anyone want to help a little?
# patch apps_makefile.patch
patching file Makefile
Hunk #1 succeeded at 41 (offset -6 lines).
Hunk #2 FAILED at 70.
1 out of 2 hunks FAILED --
I've got three 7960s running v6 SIP firmware. My Asterisk setup has
worked fine with grandstream devices, and basically, we're just
upgrading to use nicer phones.
Whilst I can make/receive calls from the 7960 to/from gossiptel).
When I try to place a call, I get the following
Jan 21 11:09:23
Is there any way to encrypt the PIN numbers in voicemail.conf.
I looked at the Wiki page for voicemail.conf but it did not mention
anything about that topic.
I am not using MySQL or any other thrid party database.
Kurt
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Hi,
We are using asterisk at the office and the incoming line is an ISDN
(HFC-PCI card with zap_hfc driver from bristuff 0.2.0 RC3a).
And I have a problem, when both ISDN B channels are in use (i.e. 2
calls in progress) it seems that anyone that calls in gets no answer
at all, and after 20
Answering my own question. The problem is that the sipura does not support
some of the sip compact header format that is now used by CVS-HEAD. I have
been sent a quick fix version of the firmware but it will be generally
available in the next firmware release.
Compact headers are not on by
I've just stumbled across a rather weird problem and was wondering if
someone could shed some light on the situation.
In testing faxing through Asterisk using Voicepulse Connect for
trunking I am able to receive faxes without a hitch. Quite impressive
considering previous experience with
On January 21, 2005 10:38 am, Nabeel Jafferali wrote:
http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
a mailing list looks like to most people, and you can see why
replying to a message, erasing its contents and starting an
entirely new email about a different topic is
bit rate is 1bps, giving 1667 bytes/sec
packetization is 20ms, giving 34 bytes per packet
Actually, iLBC in asterisk uses 30ms frames..
Everything asterisk does is in 20ms frames. In IAX and RTP this is
hardcoded, so whatever iLBC think it's doing, asterisk does 20ms.
IAX header is 4 bytes
Same here.
I called them yesterday plus email and still no reply.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, January 21, 2005 10:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Is it possible to somehow monitor/log packet loss and/or jitter in
RTP? I want to know how things look if someone complains about audio.
ethereal can do some of this for rtp, I think. At the very least, if
the endpoint supports RTCP (most do, except for asterisk), it can show
you the contents
Hi there,
we've announced a new release of our chan_misdn channel driver.
chan_misdn is a GPL channel driver for the new Linux ISDN-Layer mISDN
(www.isdn4linux.org).
So you can use all from mISDN supported ISDN catds in Asterisk.
Feel free to donwload and test it at :
Hi,
I hope this isn't a double-post...but here goes. I have setup an * box using
WBEL, and I have * up and running. The problem I have is that when I dial an
extension I cannot hear anything. It's not my sound card either. I can see the
call going through on the CLI and I see where it goes to
Thanks Brian I had forgotten that I had added that to my config!
Chris
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, January 21, 2005 4:36 PM
Subject: RE:
Andrew Kohlsmith wrote:
Thunderbird, Eudora, hell even Pine I think.
Thunderbird works very well but you have to enable it, since it doesn't do it
by default. View - Sort by - Threaded
-A.
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I want to put a single voice-mail box on a remote server, where I have
metered bandwidth. Before I do this, I want to make sure it's feasible.
Could someone confirm the following math for me?
G.711, at 64kpbs has a rated network load of 88kbps.
So for each second of conversation, about 11KB
Hi Mike -
I hope this isn't a double-post...but here goes. I have setup an * box
using WBEL, and I have * up and running. The problem I have is that
when I dial an extension I cannot hear anything. It's not my sound
card either. I can see the call going through on the CLI and I see
where it
Apply the patch manually. The changes are not that significant.
-Brian
On Fri, 21 Jan 2005 at 09:23 Mike Dewey ([EMAIL PROTECTED]) wrote:
morn all,
I am trying to compile 1.0.4 and SpanDSPpre10 to compile and have problems
patching the apps makefile. Anyone want to help a little?
#
On Fri, 2005-01-21 at 09:23 -0700, Mike Dewey wrote:
morn all,
I am trying to compile 1.0.4 and SpanDSPpre10 to compile and have problems
patching the apps makefile. Anyone want to help a little?
# patch apps_makefile.patch
patching file Makefile
Hunk #1 succeeded at 41 (offset -6
Would you please be so kind and share your config and/or phones used,
with the list so we can help you?
On Fri, 21 Jan 2005 12:07:57 -0500, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
I hope this isn't a double-post...but here goes. I have setup an * box using
WBEL, and I have * up and
Interesting, because I also called them, and I was able to get a
price, they told me $300 per license, for bulk every 5th license is
free.
On Fri, 21 Jan 2005 11:43:13 -0500, Ferguson, Michael
[EMAIL PROTECTED] wrote:
Same here.
I called them yesterday plus email and still no reply.
Hello world.
A Colombian LUG has published an article written by Diego A. Asenjo G.,
a young engineer from the also young VoIP enterprise Avatar ltda.
(http://www.avatar.com.co), about DeStar, a web frontend for Asterisk.
The article pretends to inform about this project and atract some users
What are the advantages in using mISDN over other solutions?
If I knew why it was a good idea (like does it have better sound quality than
alternatives?) then I would put the time in to test it, and also improve the
Wiki.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
At 07:22 AM 1/21/2005 -0600, you wrote:
I'm having the exact same issue on a brand new Dell Poweredge 700, using
FC2. It locks the machine totally.
-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:06 PM
To: Asterisk Users Mailing List -
Hi all
Have some bady working asterisk with oracle?
thanks in advance
wert
Do you Yahoo!?
Yahoo! Search presents - Jib Jab's 'Second Term'---BeginMessage---
Hi all
Have some bady working asterisk with oracle for the CDR?
thanks in advance
wert
---End Message---
Hello,
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
Any and all help is greatly appreciated.
The (hopefully relevant) conf file excerpts are:
Hi!
There's any way to set up a call using
G726 (sippeer)receive it on Asterisk convert it to G711Mu to send it
to PSTN broadband termination?
I've put the following in
sip.conf:
disalow=all
allow=gsm
allow=g726 (my TAs use G726
32K)
best regards,
Helder
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