I think this is what you want
http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail#comments
Let me know if you have success, I want to do something similar when I get
to that stage.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
When my service was down briefly,
On 5/18/05, Wilson Pickett [EMAIL PROTECTED] wrote:
The turnaround on email support was almost immediate.
The RMA request, once that decision was made, was easy.
I'll come back in a few days with the end of the story :)
The phone went to a test area for a day or two, the decision was
Hi all,
I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I
could not use it to run a remote stream, if I use mpg123 in command line, I
can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3
file could not be replay with asterisk.
I would appreciate with
I wonder if the combination of qualify=yes and ChanIsAvail()
does something useful? I always meant to find out. Asterisk
does seem to monitor the outbound links and does seem to be
aware when things are down when qualify is on.
It would be really useful, especially for those of us in
Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA
numbers to the asking.
-Original Message-
From: trixter http://www.0xdecafbad.com
[mailto:[EMAIL PROTECTED]
Sent: Thursday, June 02, 2005 9:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
On Fri, 2005-06-03 at 01:17 -0500, Jay Milk wrote:
Doesn't www.sipgate.co.uk do that? After all, they provide free NCFA
numbers to the asking.
You misunderstand I am asking for termination *to* NCFA. I want to be
able to call them, as my signature indicates I already have a NCFA for
inbound
Maybe you should review these:
http://asteriskdocs.org
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
I've never seen the register line you have used, the ones I see are
mostly like this:
register = username:[EMAIL
Hey, not sure if you figured it out yet or not but this is what I
came up with.
You could create a local context, an international contex and an
internal pbx-users context
make the default context of extens 300 and 301 part of the internal
pbx-users context in sip.conf or iax.conf
On Fri, 20 May 2005, Altus Snyman wrote:
A fix for what?
chan_capi-0.3.5 uses a wrong file descriptor for a pipe to asterisk and the
fix (just one line) is part of this patch as well in 0.4.0-PRE1.
I think the patch in that link is broken because I had to take out a lot
of end of lines
Hello,
im trying to connect LCS to asterisk which will act as pstn gateway for LCS.
Microsoft system supports only SIP TCP connections but asterisk UDP.
im was searching about conversion beetwen TCP and UDP and i found that SER
can do that but i don't know SER and my trying to configure SER
After a big help from Peter Svensson, I got ISDN Data-calls up and
running (will probably send a small patch to Mantis soon).
But now when everything seems connected, pppd has been authorized by
other peer and even got an IP address, the whole connection seems to
stop working.
Very unregulary,
Getting the proper version of MPG123 (v0.59r) setup under OS X was
unusually complicated for me. Sorry for pestering the list with the
MP3Player() issues I kept posting about. Your inbox will be happy to
know that I finally got the correct file at:
http://www.macupdate.com/info.php/id/6275
You misunderstand I am asking for termination *to* NCFA.
Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min
I figure if they can provide origination for NCFA numbers, they can
also terminate to them... your +44 870 number is a NCFA one, no?
--Luki
These past few days, I was struggling to create a complex (to a newbie
like me) menu, while sorting and trying out all the sound files in the
add-on package, I wondered to myself, just who is this Allison Smith
with such an angelic sound..
After a couple of minutes googling it turns out that
On Mon, 30 May 2005, Mike Price wrote:
That's what I meant with the 'CVS_HEAD setting is not good', because some
changes were made between some releases and the chan_capi Makefile just
knows 'old' and 'new' which is not working for some versions.
Anyway 1.0.7 should work with unpatched
I am trying to make 1 soft SIP UA behind NAT connect to a public hard
CISCO UA via a public asterisk server. The CISCO UA can hear the voice
from the SIP UA but not vice versa. I do set nat to yes for the soft
phone. Any help would be greatly appreciated.
Below is my sip.conf
[general]
port
Naw, I understood you full well. You'd think if they provide
origination to NCFA numbers, they'd provide termination to them as well,
wouldn't you? As far as their website is concerned, there are only two
UK rates, and no disclaimers that NCFA would be excluded.
-Original Message-
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Waldo Rubinstein wrote:
I installed Asterisk on Gentoo using emerge. At first, emerge tried
installing version 0.9 but reading the wiki showed how to get the
latest stable. I'm running Gentoo kernel 2.6.11-gentoo-r9.
Asterisk seems to be
On Fri, Jun 03, 2005 at 09:52:52AM +0200, Armin Schindler wrote:
When do you expect to have your reworked chan_capi ?
I still want to fix some race-conditions. It should be ready for
'testing'
end of this week.
Let me know so I can give it a go.
The reworked chan_capi
On Fri, 3 Jun 2005, Eric Yu-Wei Sung wrote:
I am trying to make 1 soft SIP UA behind NAT connect to a public hard
CISCO UA via a public asterisk server. The CISCO UA can hear the voice
from the SIP UA but not vice versa. I do set nat to yes for the soft
phone. Any help would be greatly
I'm trying to find a voip-suitable USB headset (I.E. headphones +
microphone) which I can use with my laptop while I'm traveling and using
Firefly or another softphone.
I'm currently using a Logitech headset which works well (except the echo
it generates toward the other caller when I turn up the
On Fri, 3 Jun 2005, Ralf Schlatterbeck wrote:
On Fri, Jun 03, 2005 at 09:52:52AM +0200, Armin Schindler wrote:
When do you expect to have your reworked chan_capi ?
I still want to fix some race-conditions. It should be ready for
'testing'
end of this week.
Let me
Hi all,
for an unknown reason, I find my asterisk server down every morning as
a result of failing to restart during the night because of a
segmentation fault. The error message is as follows:
Waiting for inactivity to perform restart...
Executing last minute cleanups
== Destroying any
Hi experts,
I wish someone would kindly give me a hand on a
problem on Asterisk Realtime.
May I know how to enable the debug messages for the
Asterisk SIP Registrar query the SIP user data in the created MySQL table. I
found that I can see the debug message for cdr_mysql which shows it can
Hey everyone here's my problem.
Have a queue configured, it plays the desired recording, checks to see if
agents are logged in via agentcallback, forwards the call according to
distribution method, times out according to timeout settings, logs out the
agent that did not answer, hunts for next
How about using a bluetooth headset? You would just need a bluetooth
dongle for the laptop to provide the wireless connection for the headset...
Mark
(i'm in the process of trying this with an old usb bluetooth dongle
(trying to find a suitable driver and manufacturers appears to have
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used
line. Would the following work for 911 calls?
[e911]
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,Dial(Zap/1/911)
exten = 911,3,Hangup()
exten = 911,102,ChanIsAvail(Zap/4)
exten = 911,103,Dial(Zap/4/911)
exten =
Mojo with Horan Company, LLC a écrit :
While I agree with Firefly being a top-notch IAX client, Hendrik was
hoping for a linux client. I'm also curious which one people recommend.
Kiax, iaxcomm, mozphone.
Thanks!
Michael Graves wrote:
On Thu, 2 Jun 2005 12:51:23 +0200, Hendrik Wouters
On Fri, 2005-06-03 at 00:10 -0700, Luki wrote:
You misunderstand I am asking for termination *to* NCFA.
Can you also terminate through Sipgate? They say: United Kingdom, 1.19 p/min
I figure if they can provide origination for NCFA numbers, they can
also terminate to them... your +44 870
On Fri, 2005-06-03 at 03:33 -0500, Jay Milk wrote:
Naw, I understood you full well. You'd think if they provide
origination to NCFA numbers, they'd provide termination to them as well,
wouldn't you? As far as their website is concerned, there are only two
UK rates, and no disclaimers that
Hello,
we got a SNOM 360 here and this gota TRANSFER button.
With this i can transfer a call from my phone another one. But when i
push this Button and transfer the call to another phone, i get kicked out.
Now, every secretary first asks the chief if he is available or not -
how can i
So I'm on the phone with an Army office in Iraq as part of my other
paying job (I'm a radio journalist) and we got to talking about the
technology I was using at my end to record our conversation (I called
him from my * box) when to my total amazement he announced that they
were running
Hi.
Agree with the above, you probably need the crc_citt module. Modules
should be loaded before starting zaptel. Use ztcfg -v to get specific
error messages. If your using this in a production environment I
suggest buying a card from digium to use as a timer, I tried using
ztdummy but
Sure Adrian.
Im working on putting together a howto for this.
Ill post a message when its done.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Adnan Ahmed
|Sent: Jueves, 02 de Junio de 2005 12:46 a.m.
|To: Asterisk Users Mailing List -
Hi Waldo.
Don't worry about it. We have 13 sites scattered evenly around the
globe, latency up to 700 ms. This is of course noticeable, but I found
that for most cases we get better quality than using PSTN between our
sites, reason probably being that Telcos are often using low-quality
Simple call to any female for this table plase...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Adam Vocks
|Sent: Jueves, 02 de Junio de 2005 02:50 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] a
Hehehe then something is weird with mine :=
I have this
Zapata.conf
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800
And now:
voip*CLI zap show channel 1
Channel: 1
File Descriptor: 18
Span: 1
Extension:
Dialing: no
Context: casa
Caller ID:
Calling TON: 0
Caller ID
Hi there
I am in the UK.. and am running latest asterisk on FC1 (2.4 kernel). I would
like to know what the best option is for a 4 port BRI card. I notice Digium
don't provide one.. I have heard the Junghanns do one...but are there others
??
Is the Junghanns card reliable/stable with good sound
Hello,
We use Asterisk with PAP2 and today we connected the FXS ports of PAP2
to CO ports of our Panasonic KX-TD1232. Problem is that Panasonic
doesn't ring - that is doesn't ring every time the PAP2 is ringing.
When we reset either Asterisk or the PAP2 it usually rings, but after
couple of
Hi,
Eicon Diva 4BRI Card and chan_capi.
--- Brett, Gary [EMAIL PROTECTED] wrote:
Hi there
I am in the UK.. and am running latest asterisk on
FC1 (2.4 kernel). I would
like to know what the best option is for a 4 port
BRI card. I notice Digium
don't provide one.. I have heard the
I wonder if the combination of qualify=yes and ChanIsAvail()
does something useful? I always meant to find out. Asterisk
does seem to monitor the outbound links and does seem to be
aware when things are down when qualify is on.
It would be really useful, especially for those of us
Or the sirrix,I think its the cheapest and there was lots of dev. on the
drivers
On Fri, 2005-06-03 at 04:49 -0700, Nardis Dome wrote:
Hi,
Eicon Diva 4BRI Card and chan_capi.
--- Brett, Gary [EMAIL PROTECTED] wrote:
Hi there
I am in the UK.. and am running latest asterisk on
Hello,
I am pretty new with Asterisk and I am using it as an H323 gateway.I
would like to keep the same h323-conf-id in the outgoing leg as in the
incoming leg.
So far I have only tried inaccessnetworks' oh323 module, but I think
this is a more generic issue. My extensions.conf is pretty simple:
Wow that eicon is certainly quite expensive... ive found 2 sellers in the UK
so far selling at over £1000 . I think the Junghans comes in at around £600.
Is the Eicon that much better ?
-Original Message-
From: Nardis Dome [mailto:[EMAIL PROTECTED]
Sent: 03 June 2005 12:49
To: Asterisk
It also fails.
# /etc/init.d/zaptel start
* Starting zaptel...
Notice: Configuration file is /etc/zaptel.conf
line 206: Unable to open master device '/dev/zap/
ctl' [ ok ]
# lsmod
Module Size Used by
Any other ideas?
Thanks,
Waldo
On Jun 3, 2005, at
Hi!Does anyone have any ideas on how to build an interactive IVR where questions are asked by Asterisk (pre-recorded prompts), the caller answers the questions, and the system records the answers and emails the whole question-answer session as a .wav file? Similar to Comedian Mail except an menu
Have you read and executed the instructions in
/usr/src/zaptel/README.udev
It also fails.
# /etc/init.d/zaptel start
* Starting zaptel...
Notice: Configuration file is /etc/zaptel.conf
line 206: Unable to open master device '/dev/zap/
ctl'
Good morning!
Sorry if this gets posted twice, I tried via NNTP before, and it didn't
seem to work.
After having downloaded the tarball for Asterisk addons via FTP from
ftp.asterisk.org I am trying to install it following the instructions on
With sipura (I tried this with both the 3000 and 841) set to prefer
the g726-32 codec, a call from the sipura to asterisk will use g726.
Asterisk sip.conf has:
disallow=all
allow=g726
allow=gsm
allow=alaw
When the call is from asterisk to the sipura, asterisk will not use
g726. It
Does anyone have any ideas on how to build an interactive IVR where
questions are asked by Asterisk (pre-recorded prompts), the caller answers
the questions, and the system records the answers and emails the whole
question-answer session as a .wav file? Similar to Comedian Mail except an
You haven't upgraded your kernel since you installed the zaptel package, did
you?
If you do:
ls /lib/modules/kernel version/misc
do you see ztdummy.ko in there?
I have a system running 2.6.11-gentoo-r6 and zaptel 1.0.7 with no problems.
However, I load the modules in
I'd settle for a check that the host was not overly lagged - most of our
problems come from internet outages. I'll take my chances that the provider
will function.
Chris Mason
www.anguillaguide.com
Tel: (305) 704-7249 Fax: (815)301-9759
___
Garth Brown wrote:
I have my Asterisk server all setup. But have an odd problem and hope
someone here can help.
I have a Polycom IP 300, a Grandstream GXP-2000, and an installation
of X-Lite. They can each call each other just fine
(extension-to-extension). I can also dial-in from the
Ok. I recompiled the kernel with CITT support.
After rebooting, I do:
# modprobe zaptel
# dmesg | tail
Zapata Telephony Interface Registered on major 196
Then, I do:
# modprobe ztdummy
FATAL: Error inserting ztdummy (/lib/modules/2.6.11-gentoo-r9/misc/
ztdummy.ko): No such device
# dmesg |
I just added all the changes to /etc/udev/* as specified in
README.udev and still it does not work. I still get the same results
as in my previous post :(
Help please
Thanks,
Waldo
On Jun 3, 2005, at 9:25 AM, Rich Adamson wrote:
Have you read and executed the instructions in
I haven't upgraded my kernel. I have the same kernel.
The only difference I have now is I didn't have anything in
modules.autoload.d. So I made the changes and rebooted.
# lsmod
Module Size Used by
zaptel180708 0
# modprobe ztdummy
FATAL: Error inserting
What about some sort of asterisk-level Ping app that could let one
server with the app, ping the other, and check for status info, and if
it doesn't like what it sees (or doesn't see anything), it would
consider that channel dead?
I know I'm just passing broad strokes here, but I think the idea
I am wondering if the SIP protocol and its implementation in * allows for
changing codecs mid-connection.
I've seen some questions regarding this on the list, but I've not found any
clear answers.
I've also seen the SIP_CODEC variable, but it's not clear that it will change
the codec on an
--- Brett, Gary [EMAIL PROTECTED] wrote:
Is the Eicon that much better ?
sorry, i have only experience with Eicon... maybe
someone else is able to give a feedback...
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection
Ok. I re-ran emerge after recompiling the kernel with CCIT support
and left everything as you have it in your system and now it all
works just fine.
Thank you all for helping me.
Waldo
On Jun 3, 2005, at 9:14 AM, Waldo Rubinstein wrote:
I haven't upgraded my kernel. I have the same
If you plan to go this route don't buy a bluetooth adaptor that uses the
XTNDconntect software. I've never been able to get it to work properly
and there are no updates since last year (from the hardware vendor at
least). Its an Innovision Wavelinker USB bluetooth module. I can
discover and
How do I match by username instead of by host/ip? By default this is how
it should work, but it does not. we do not have insecure turned on.
Matt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Friday 03 June 2005 14:28, Nardis Dome wrote:
--- Brett, Gary [EMAIL PROTECTED] wrote:
Is the Eicon that much better ?
sorry, i have only experience with Eicon... maybe
someone else is able to give a feedback...
Aside from paying for a recognised brand name, with Eicon you get on-board
I'm going to try and ask this again and keep it short and as too the
point as I can while still providing enough info to be of use.
PLEASE advise if I am going about this wrong or asking too much.
I'm seriously doing my BEST to throughly read the docs and try a bunch of
things BEFORE coming
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used
line. Would the following work for 911 calls?
[e911]
exten = 911,1,ChanIsAvail(Zap/1)
exten = 911,2,Dial(Zap/1/911)
exten = 911,3,Hangup()
exten = 911,102,ChanIsAvail(Zap/4)
exten = 911,103,Dial(Zap/4/911)
exten
When your TDM card does not load.
Plug in the molex power tap. I helps :)
10 hours wasted on downloading different Zaptel and Asterisk versions
and the I must be stupid feeling to find it unpluged in the
computer..
--
sig
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
You are looking for consultative Xfer and attempting a blind one. Gotta
put the first call on hold first and then join it with the second (line
to boss) using the Xfer key.
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christian Hiller
Sent:
Hi, Im using an Asterisk Server and a Cisco AS5350. They
are interconnected via Sip. When I tried using G.729 codec, all recorded
announcement of asterisk is no longer heard in the system but when I bring it
back to G.711 the announcement works perfectly.
Any idea how I can make the
On 6/3/05, Asterisk User [EMAIL PROTECTED] wrote:
Hi experts,
I wish someone would kindly give me a hand on a problem on Asterisk
Realtime.
May I know how to enable the debug messages for the Asterisk SIP Registrar
query the SIP user data in the created MySQL table. I found that
Just a slight correction, in case the OP didn't realise it was an error:
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
When dialing an outbound sip call (via your sip provider), the Dial()
statement can use the form:
exten = _1XX,1,Dial(SIP/myOutContext)
[EMAIL PROTECTED] 1.1 Released and can be downloaded from Sourceforge.
http://sourceforge.net/project/showfiles.php?group_id=123387package_id=135368
Cheers,
Max W. Blackmer
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Fri, 2005-06-03 at 06:28 -0700, Nardis Dome wrote:
--- Brett, Gary [EMAIL PROTECTED] wrote:
Is the Eicon that much better ?
sorry, i have only experience with Eicon... maybe
someone else is able to give a feedback...
I'm using Junghanns 4 port card. There is also an 8 port card.
Christian,
I don't have any specific answers about your particular SNOM device, but
what you are wanting to accomplish is an attended transfer, instead of a
blind-transfer. You should verify that the SNOM 360 is capabile of
doing an attended transfer. Cisco 79xx series phones provide both
How do I match by username instead of by host/ip? By default this is how
it should work, but it does not. we do not have insecure turned on.
Its my understanding (which could be very wrong) the current sip
implementation within asterisk has a number of limits, and that
appears to be one of
Eric,
The problem you are seeing is because the RTP (voice) packets being sent
towards the NAT'd UA are being blocked by the NAT router.
The UA being used behind NAT will need to have a static IP address set
(e.g. 192.168.1.50) and on the NAT router you will need to permanently
forward
Try this:
1) You're on a call
2) Push a Line button, so that you get dialtone
3) Dial the boss extension #
4) Hey boss, you have a call from XXX
5) Push Transfer
6) You can select which call to transfer (if you have more that 1 on hold)
7) Push transfer again.
Julian.
On 6/3/05, Christian
Hey, everybody---
Ignorance CAN be bliss, at least for a while, but, Just so you
know...
A week or two ago, some upgrades to the expression parser (you know, the
expressions you put in $[ ... ] in your extensions.conf file) that I
submitted, have been merged into the CVS HEAD of the
Hmmm, the sirrix isn't even that much cheaper than the Junghanns boards.
The difference is only EUR 100
On Fri, 3 Jun 2005, altus wrote:
Or the sirrix,I think its the cheapest and there was lots of dev. on the
drivers
On Fri, 2005-06-03 at 04:49 -0700, Nardis Dome wrote:
Hi,
Eicon Diva
Hi.
I want to handle incoming chan_misdn traffic by asterisk, but I've got
message - 'Extension can never match, so disconnecting'. What I'm doing
wrong ? How I can pass incoming dialed number (dad) to misdn context (in
my case 'dss1_incoming') ? Works unrouted calls (s extension) if I
On Friday 03 June 2005 01:36, Chris Coulthurst wrote:
Any suggestions? Dialplan examples?
Yeah; don't post this kind of message to this list unless you've verified it's
a problem. There are NUMEROUS places online to check routing to a host.
Personally I use dnstools.com.
Perhaps this
I'm using Junghanns 4 port card. There is also an 8 port card.
Installation is very simple, download a startup image from Junghanns.net
and it does the rest... It works - I've no complaints.
ps - FAX reception works - as part of asterisk.
I assume you are using spandsp?
I am thinking of
Just a slight correction, in case the OP didn't realise it was an error:
In article [EMAIL PROTECTED],
Rich Adamson [EMAIL PROTECTED] wrote:
When dialing an outbound sip call (via your sip provider), the Dial()
statement can use the form:
exten =
On Friday 03 June 2005 15:19, Remco Barende wrote:
Hmmm, the sirrix isn't even that much cheaper than the Junghanns boards.
The difference is only EUR 100
Telephony is an expensive game to be in :) But my ISDN card was only 20 EUR!
suddenly doesn't mean an awful lot when you have to
What does this mean? I have a sipura 3000 with an analog line that I
have created as a trunk. Incoming calls make it to the sipura but not
to the pbx. However I can make outgoing calls but have no audio. I
thought it might be a codec issue so I set disallow=blank and
commented out the allow=.
That's great.it's a virus I tell you * is everywhere :)
Viva la asterisk.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Friday, 3 June 2005 6:46 AM
To: Asterisk Users Mailing List -
On Fri, 3 Jun 2005, Remco Barende wrote:
I am thinking of another solution for fax. I have an * box with one PRI
card and I'm thinking of adding a quad BRI card in the same box.
A separate box running fasx server software will also be equipped with a
BRI card for faxing (I cannot use
If Zap/5 is the least-used line, dial that one first :) Other than
that, you could use a dial-group as someone else suggested.
-Original Message-
From: Chris Coulthurst [mailto:[EMAIL PROTECTED]
Sent: Friday, June 03, 2005 4:51 AM
To: Asterisk Users Mailing List - Non-Commercial
Ahhh... Sneaky. Because of the special billing agreements on NCFA
numbers, there's bound to be a lower limit to how these calls are
priced. I doubt BT gives sipgate (or any other VOIP provider) a
signigicant discount on these calls. If you can reasonably expect that
there are a lot of other
Did you install G729 codec and changed sip.conf accordingly? Or is it
just announcements?
On Fri, 2005-06-03 at 21:59 +0800, Nathaniel Angelo A. Torres (247talk)
wrote:
Hi, Im using an Asterisk Server and a Cisco AS5350. They are
interconnected via Sip. When I tried using G.729 codec, all
I modified sip.conf to allow g729. The call is getting through but the
announcement is not.
Any idea?
Thanks.
Cheers,
nat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Soner Tari
Sent: Friday, June 03, 2005 11:07 PM
To: Asterisk Users Mailing List -
I contacted SNOM and they told me to change a
couple of options but still no lights, here is what they told me
Line page SIP tab:o Long SIP-Contact (RFC3840) to "off"o
Support broken Registrar to "on"Advanced page:o Filter Packets
from Registrar to "off"
And please ask the Asterisk
Hi Remco,
-Original Message-
I am thinking of another solution for fax. I have an * box
with one PRI
card and I'm thinking of adding a quad BRI card in the same box.
A separate box running fasx server software will also be
equipped with a
BRI card for faxing (I cannot use
What does this mean? I have a sipura 3000 with an analog line that I
have created as a trunk. Incoming calls make it to the sipura but not
to the pbx. However I can make outgoing calls but have no audio. I
thought it might be a codec issue so I set disallow=blank and
commented out the
On Fri, 2005-06-03 at 10:14 -0500, Jay Milk wrote:
Ahhh... Sneaky. Because of the special billing agreements on NCFA
numbers, there's bound to be a lower limit to how these calls are
priced. I doubt BT gives sipgate (or any other VOIP provider) a
signigicant discount on these calls. If you
You need to install G729 codec on *, because announcements need it (not
pass-thru).
Check if it's installed with 'show translation' on CLI, you should see
some values not -'s.
If not, install from here http://www.readytechnology.co.uk/open/g729/
See http://www.voip-info.org/wiki-ITU+G.729 also
Here is another one.
http://vanabel.com/
On 6/3/05, Mark Phillips [EMAIL PROTECTED] wrote:
So I'm on the phone with an Army office in Iraq as part of my other
paying job (I'm a radio journalist) and we got to talking about the
technology I was using at my end to record our conversation (I
Hi i'm trying to connect to the PSTN in the
following way
sip ATA - * - gnugk - Cisco AS5300
- PSTN
I'm using asterisk CVS-HEAD-06/01/05-14:33:15
running over RH EL3
Asterisk-Oh323 0.7.2 pre1
Open H323 v1.13.5
pwlib v1.6.6
and I'm having a lot of trouble, gnugk and * both
have public
Christian Hiller wrote:
Hello,
we got a SNOM 360 here and this gota TRANSFER button.
With this i can transfer a call from my phone another one. But when i
push this Button and transfer the call to another phone, i get kicked
out.
Now, every secretary first asks the chief if he is available
I wonder what this means:
WARNING[16206] app_voicemail.c: No origtime?!
WARNING[16206] app_voicemail.c: No origtime?!
Running cvs head, I see these in the * logs.
Any one have tips on this error?
--
respectfully, Joseph ===
-= ** =
Hi,
I have a very stupid PABX that works ok when receiving calls, but
bad when originating them. Asterisk complains a lot (hundreds of log
lines) abouth 'VAD' and no audio is heard.
I am using -stable. Is there another newer version that works
and supports VAD better?
thanks.
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