Anyhow, Asterisk1 and Asterisk2 are connected using IAX2.
What I would like is to have the SPA3000 Message Waiting indicator
based on the voicemail message hosted on the Asterisk2 server.
There is this old patch that does remote MWI over IAX (among other
things). I used it on earlier versions
Hi Dan,
Thanks for the reply.
I was just searching the MII china site to confirm
this. But could not find any information on the same.
They just kept talking about issuing VOIP licences to
service providers But no focus on whether an
enterprise can do it for itself or not.
But just one point, Do
Hello,
is it possible to restart the wct4xxp kernel module and start again
without stopping Asterisk?
i tried to unload chan_zap.so but rmmod says the module is in use.
Is it possible? if it is, how its possible?
Thanks
Nico
___
--Bandwidth
On Thu, 2006-07-27 at 09:04 +0200, [EMAIL PROTECTED] wrote:
is it possible to restart the wct4xxp kernel module and start again
without stopping Asterisk?
Yes, you should be able to unload chan_zap.so from Asterisk without
stopping the rest.
i tried to unload chan_zap.so but rmmod says the
- Message d'origine
De: Mojo with Horan Company, LLC [EMAIL PROTECTED]
A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 22:39
Yes, on a Zap FXO channel,
I didn't want to send the Agent thru the whoule AgentCallbackLogin
rutine just to _log off_.
This does not make really sense to me.
thank for your answer anyway.
Kai
Here is what I do...
Exten=777,1,AgentCallbackLogin()
___
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Dear Gurus,
I'm newbe in Asterisk and I want to evaluate the system.
I have several location branch office and I want to use VOIP between them.
Is there any documentation about Asterisk that cover several location
and the dial plan?
Is it possible to have one central Asterisk to control all the
As i writed, i do an unload of chan_zap.so but i can't unload the module
wct4xxp, is this possible?
Thanks
Nico
On Thu, 27 Jul 2006, Russell Bryant wrote:
On Thu, 2006-07-27 at 09:04 +0200, [EMAIL PROTECTED] wrote:
is it possible to restart the wct4xxp kernel module and start again
Hi,
we are having some trouble with CDR records. Example:
Case 1: Customer 12345 calls extension 10. Extension 20 takes the call
using Pickup (e.g. *810). I now have two CDRs:
1: 12345 - 10
2: 20 - *810
I could live with the second CDR but the first gives the impression as
if 12345 was talking
Please any one knows from where I can download asterisk French
sounds /var/lib/asterisk/sounds.
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without
Dear Anthony,
The dial plan is currently very simple it should pick up any call
and send it to a sip phone registered, you can see the context below named
zap-in is what I am using, it is only that and nothing more, is there something
extra I have to add to dial plan or to that context ?
Thx
MAG
Hi all !
I am planing to set up around 20 SIP Phones which will be purchased in
one bunch, I am more or
less free of choice.
I wonder if anyone knows sip phones which allow configuration upon
login. The following scenario:
User logs into any phone and the settings of the phone are always the
Hello
Just use Snom or grandstream phones. They can be provisioned very easily via
HTTP. You just setup a config URL on the phones, and they get their
configurations from there. If you want to get more advanced, they can send
along their MAC address, and thereby enabling you to custom config
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Telles
Rodrigo P. Telles wrote:
Hi,
I'd like to know if some one knows how to make Asterisk record a call after
xfer (not bxfer).
I tried some ways but it doesn't work at all.
I assume that you are using Asterisk 1.0.X. - I had the same
Hi Nik,
I like the Grandstream Budge Tone 102 VoIP Phones which you can find here:
http://www.voipsupply.com/product_info.php?products_id=40
and here:
http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-31609737728.htm
Also, the GXP-2000 is a very popular model too, although once
Hello Khaled,
Follow this link:
http://svn.digium.com/view/asterisk/sounds/fr/trunk/?rev=34575
Best regards,
Olivier S.
Khaled Chehab a écrit :
Please any one knows from where I can download asterisk French sounds
/var/lib/asterisk/sounds.
//Regards//
Dear All,
I'm looking for a mobile SIP client to use with Asterisk.
Has anyone got experience in this area and can you advise me of a
product?
Many Thanks
Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management plc
SIP: [EMAIL
Hi all,
i have following setup
[]--[asterisk]--[oh323]--[HiPath]--[8000]
is my voicemail access
exten = ,1,Answer()
exten = ,2,VoiceMailMain()
8000 is an Optiset phone registered on the HiPath.
When 8000 calls i have no voice (depends on the
setting of FastStart). When
Hi
I want to use Asterisk as a Voicemail Box for my Callmanager Users
The Link between Cisco Callmanager and Asterisk has to be SIP (according
to
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration)
The Voicemail Part on Asterisk is running perfect via a IAX Softphone
Hi !
Also, the GXP-2000 is a very popular model too, although once you
consider the capabilities of Asterisk the only real advantage this unit
has over the others (even in an office environment), is the Power over
Ethernet (PoE) feature:
which is supported be Snoom as well.
Anyway I would
Has anyone else had problems with the Nokia E61 and E70 phones not
always answering voip calls?
We have them connected via a local access point (so no router/NAT) and
sometimes the phones dont ring when called. They are registered ok and
if you use the phone to make a voip call it works fine.
The
Hi,
Quickie ... is the alcatel ip touch 4068 (or any other in that series)
sip enabled?
If not, does alcatel have a sip-enabled phone?
Cesc
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On 07/27/06 03:28 Phil Jordan said the following:
Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 160 sample
intervals
Jul 26 20:05:22 DEBUG[16371] channel.c: Avoiding initial deadlock for
'IAX2/phil-5'
Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Called IAX2/phil
Jul 26
On 7/27/06, Gareth Blades [EMAIL PROTECTED] wrote:
Has anyone else had problems with the Nokia E61 and E70 phones not
always answering voip calls?
We have them connected via a local access point (so no router/NAT) and
sometimes the phones dont ring when called. They are registered ok and
if you
On 7/27/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello
Just use Snom or grandstream phones. They can be provisioned very easily via
HTTP.
You just setup a config URL on the phones, and they get their configurations
from there.
If you want to get more advanced, they can send along their
Title: Manager interface
This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on.
Regards
Lee
Hi Dave
The problem is with the way in which Asterisk handles 'overlap' dialling
with SIP. i.e. not very well at all. If you remove the early dial
feature from the phone I think you will find it will solve the problem.
The issue is that Asterisk does not apply the digit timeout on SIP early
Title: Manager interface
If you want to do a screen popup when an
agent receives a call, then you should consider looking at these events:
AgentCalled
AgentConnect
AgentComplete
p.s: I'm not sure, but you might need to set
eventmemberstatus=yes in your queue.conf to receive these
Hi all,
I'm having major trouble with
a simple asterisk installation dropping calls in the middle of the
conversation.
I recentlyupgraded from
asterisk-1.2.3 and zaptel-1.2.2 to asterisk 1.2.10 and zaptel-1.2.7, but to no
avail. The machine is equiped with a TDM40B and a
TDM22B and
I use CDR(userfield) to store dialed numbers
[call9]
exten = s,2,Set(CDR(userfield)=${number})
exten = s,3,Dial(Zap/2/${number}|40)
[outgoing]
exten = _9.,1,Set(number=${EXTEN:1})
exten = _9.,2,Goto(call9|s|1))
in cdr i have
| dst |userfield
+-+-
| s | ${number}
I
Ciao Benchev,
Also register= can be done only from a .conf file.
Well, I'm experimenting right now with this, and I can tell you that
register = works even with static realtime.
HTH,
--
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
___
Hi All
Just wondering if anyone knows of a solution to the squashed tiff
problem with spandsp (or rather Windows Image Viewer) other than
converting to a PDF. I find the PDF image quality is not nearly as good
as the original TIFF. Apparently the Windows Image Viewer doesn't
understand the
Sangoma 104D on a DL320, 3ghz, 1gig ram, NFAS, 50% CPU utilization @ 95
calls. Only running asterisk and passing calls off via ulaw SIP.
Thanks,
Steve Totaro
-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 26, 2006 9:17 PM
To: Asterisk Users Mailing
Hi All,I have a problem and I'm not sure if a solution is possible without using the asterisk testing code.I am developing a volunteer translation service that users can dial into. I have a list of volunteer translators cell phone numbers stored in a mysql database along with times that they have
Hi Friends, I am Chandra from India. Thank you for your cooperation and for clear my doubts. Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. When I am making calls
First
zaptel.conf
fxsks=1-4
zapata.conf
channel = 1-4
extensions.conf
[tutorial]
exten = s,1,Dial(SIP/350,30)
- do You have SIP/350 ? ther is onlu 300 in sip.conf
Hi Friends,
I am Chandra from India. Thank you for your cooperation and for clear
my doubts.
Now, I have installed
well, first thing, turn on debug logging in logger.conf (edit the
messages line so that it includes the word debug, the file has
examples to help you).
then, after doing a logger reload, you will be getting quite a bit of
logging...the next time a call drops, note as much about it as you can
Hi Group!Hi Wagner!Thanks for the interest. I'm from Colombia and I'm trying to develop VoIP as you know on *. So thanks again for the offering in Brazil, althought you can help me with some idea by this way.
To make the call I'm using SJphone (softphones) to make the tests. I'm not using IP
Has anyone used the new 3102? If so, does it work correctly?
I heard lots of horror stories about the SPA-3000 causing terrible echo,
picking up voice tones as DTMF, etc, so Im a little hesitant to buy.
Thanks,
Wes Baehr
--
No virus found in this outgoing message.
Checked
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Shad Mortazavi
Sent: Thursday, July 27, 2006 4:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Mobile SIP Client
Dear All,
I'm looking for a mobile SIP client to use with
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file extensions.conf but not the
correct group I configured so I changed it and will test again.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
The dial plan is
I opened bug
#0007490 the other day. The issue was that when you do a 'sip debug' on the
Asterisk console, there was no way to have this output go _only_ to the messages
file. Someone with the id of 'russell' in his infinite wisdom has deemed that
this isn't a bug, closed it, and given me
On Thursday 27 July 2006 10:32, Douglas Garstang wrote:
It clearly is a bug, or at the VERY least, a limitation that needs to be
fixed. So why the hell did he give me -2 karma points and say 'not actually
a bug'. Fine... so how do you file an enhancement request then? If there's
no way to file
I hate to say this but you might just have hit a 'reap what you sow'
moment - you don't hesitate to trash Asterisk on this mailing list
when you can't make it do what you think it should do, and just maybe
this affects how the developers treat requests from you on the bug
tracker?
Just a
Dear All,
I just wanted to comment on this point of the discussion:
> In a PRODUCTION environment, you can't be running a sip debug to your
> console.
In a PRODUCTION environment you have all of these issues
worked out in your
test lab before deploying to production.
I do agree with Douglas
Hello to all
can someone recommend me a nice SIP client with video for windows??
I tried X-Lite 3.0 but it's a lousy piece of software.
Does someone knows about a better software?
Regards
Joao Pereira
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-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 27, 2006 8:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] bugs.digium.com
On Thursday 27 July 2006 10:32, Douglas Garstang wrote:
It clearly is a bug, or at the
Garth van Sittert wrote:
Hi All
Just wondering if anyone knows of a solution to the squashed tiff
problem with spandsp (or rather Windows Image Viewer) other than
converting to a PDF. I find the PDF image quality is not nearly as
good as the original TIFF. Apparently the Windows Image
Thank you for the information.
I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek
9000.
Many Thanks
Shad
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What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free.On 7/27/06, Joao Pereira
[EMAIL PROTECTED] wrote:Hello to all
can someone recommend me a nice SIP client with video for windows??I tried X-Lite 3.0 but it's a
On Thursday 27 July 2006 15:07, Andrea Spadaccini wrote:
Ciao Benchev,
Also register= can be done only from a .conf file.
Well, I'm experimenting right now with this, and I can tell you that
register = works even with static realtime.
Not even, it *must* work because if one uses
realtime
Michael Graves wrote:
On Tue, 25 Jul 2006 19:43:58 +0100, Tim Panton wrote:
On 25 Jul 2006, at 16:23, Stephen Bosch wrote:
What are the best IAX2 hard phones?
I've got a couple of IAX hardphones, with PA168, they are useable,
but only just. They are hard to hang up (which is a design
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Shad Mortazavi
Sent: Thursday, July 27, 2006 10:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Mobile SIP Client
Thank you for the information.
I'm specifically looking
Hi Mike, Hi all,
really works. ;-)
But that can not be the solution for the future? :-) Can it?
I think there should be an ANSWER() implimented in the Transfer function to
prevent this problem ...
Or does anybody have other ideas?
greetings,
Martin
- Original Message -
From: Mike
Hi Shad,
If you haven't committed to your handsets already, you might want to
wait it out a few weeks for the HTC Hermes.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Shad Mortazavi
Sent: Thursday, 27 July 2006
I'm having major trouble with a simple asterisk installation dropping calls
in the middle of the conversation.
I was having similar issues when we installed a new non-pri T1, one of the
problems had to do with the wiring job that was done, but the major problem
seemed to be related to IRQ
- Original Message -
From: Jason Kim
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thu,
27 Jul 2006 01:51:19 -0300
Subject: [asterisk-users] DTMF relay
Hi,
My environment is ITSP---Asterisk--SipPhone.
I want to send dtmf from SipPhone to ISTP using 'info'
- Original Message -
From: Jean-Yves Avenard
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Thu, 27 Jul 2006 02:07:56 -0300
Subject: Re: [asterisk-users] Message
waiting question...
Hi
On 7/27/06, Joshua Colp
Hi all,
Hiya.
After some more thought and investigation, I think the following is
definitely my problem:
-
Looking for 10 in Outgoing
Reliably Transmitting (NAT):
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
-
Indeed.
Pure guess without looking at the trace yet, its likely to be a timing
issue (involving the codec translations) as opposed to the codec itself.
I've had good luck running g726 on iax links.
Joseph Love wrote:
Well, you win, it's definitely codec related.
Switching to ulaw causes this issue
On Jul 27, 2006, at 7:32 AM, Douglas Garstang wrote: I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed
Dashy wrote-
Hi Dan,
Thanks for the reply.
I was just searching the MII china site to confirm
this. But could not find any information on the same.
They just kept talking about issuing VOIP licences to
service providers But no focus on whether an
enterprise can do it for itself or not.
2006/7/27, Cesc [EMAIL PROTECTED]:
Hi,Quickie ... is the alcatel ip touch 4068 (or any other in that series)sip enabled?If not, does alcatel have a sip-enabled phone?CescNo (for both questions), as SIP is seen as a low end protocol, yet unable to transport high end features of ip touch phones.
Have you tried setting:
--
Faxdetect=no
Busydetect=no
Callprogress=no
Busycount=8
--
In zapata.conf?
John Coleman - IT Specialist
SunWest Education Credit Union
http://www.swecu.com
well, first thing, turn on debug logging in logger.conf (edit
On Thu, 2006-07-27 at 09:40 +0200, [EMAIL PROTECTED] wrote:
As i writed, i do an unload of chan_zap.so but i can't unload the module
wct4xxp, is this possible?
Well, you actually said that the error was for chan_zap.so. :)
Anyway, you should never need to unload the wct4xxp driver. If you
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Joshua Colp wrote:
Hi all,
Hiya.
After some more thought and investigation, I think the following is
definitely my problem:
-
Looking for 10 in Outgoing
Reliably Transmitting (NAT):
SIP/2.0 484 Address
- Original Message -
From: Dave Hope
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Thu, 27 Jul 2006 14:46:02 -0300
Subject: Re: [asterisk-users] SIP Woes
Thanks for the suggestion, I added that in and now get:
Hi all, still trying to debug this Goldmine CRM softphone, it does appear
that the client is being authenticated, but the server is replying with this
message (in the /var/log/asterisk/full file)
DEBUG[4307] chan_sip.c: SIP message could not be handled, bad request:
[EMAIL PROTECTED]
Does that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Joshua Colp wrote:
- Original Message -
From: Dave Hope
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Thu, 27 Jul 2006 14:46:02 -0300
Subject: Re: [asterisk-users]
Hi all, has anyone EVER seen a SIP phone that has a 'data port', like most
business phones do? We use this in our office to connect analog wireless
headsets to the desk phones (plantronics ct12s) and need to continue using
these (as we have a significant amount of $$$ invested in them). I've
Hi, i'm having problems with DTMF, the problems are with established
connections and some IVRS.
When i call to other number which has an IVR, some digits doesn't
work. I digit a long number (required by the IVR, at least a 10 digit
number) and it doesn't work. I think it's about DTMF signalling,
Hi all, still trying to debug this Goldmine CRM softphone, it does appear
that the client is being authenticated, but the server is replying with
this message (in the /var/log/asterisk/full file)
DEBUG[4307] chan_sip.c: SIP message could not be handled, bad request:
[EMAIL PROTECTED]
Just turned
I was just wondering if anyone out there has tried vitelity for
VoIP service If you did what is your story with how good/bad they are?
Thanks!
Curt
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The subject pretty much describes what I need to do.
Basically, I want to be able to detect that there is voicemail waiting at
the CO on an FXO port and somehow flash the message waiting light on an
H.323 phone (or any other type of phone)
I my case, the CO is actually a legacy POTS based
On Thu, 2006-07-27 at 15:36 -0400, Curt Shaffer wrote:
I was just wondering if anyone out there has tried vitelity for VoIP
service If you did what is your story with how good/bad they are?
I have just recently made the switch from Sixtel to them (because
Vitelity bought Sixtel). For
I have used exgn for several months and after the merger I have had now problems.On 7/27/06, Curt Shaffer [EMAIL PROTECTED]
wrote:
I was just wondering if anyone out there has tried vitelity for
VoIP service If you did what is your story with how good/bad they are?
Thanks!
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey all,
I have a x86 Pentium D asterisk system with two Digium 400's in it. I am
establishing a IAX2 Connection to another Asterisk system running on a
Solaris server.
When a call is placed between the two systems, everything seems fine for
a
Can anyone tell me how to configure the grandstream gxp-2000 for 4 line
apearances. I have the the sample conf from the website and the phone
is getting its config from my TFTP server. But it does not have any info
for the other line apearance butons
The real thing that would help is a complete
Hello,
Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/
Asterisk. I have pointed a DID from my carrier via SIP through g729 to
asterisk. Problem is I am not getting any audio even though I am getting
DTMF in asterisk. I am trying to run A2billing with asterisks.
On 27 Jul 2006, at 11:47, Lee Archer wrote:
This has probably been discussed before but I need to do a screen
pop and I'm looking for ways to do it. I am assuming I need to use
the manager interface, which is ok cos I'm using that for calling
out but I'm not quite what to pick up on.
I am with Vitelity now, they just completed merging with EXGN which i was
signed up with. I also signed up with them with some of my clients. So far
(last 4months) no issues what so ever. Great service, and Customer support
is timely and very knowledgeable.
Terrelle
From: Curt Shaffer
On 7/26/06, Zenone [EMAIL PROTECTED] wrote:
But my question was, is it possible to free the channel if it rings toolong?MichelUsing this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls ().
--
Two questions:
We need to run Ethernet out to a really long distance
20,000ft. We have the ability to put a powered repeater in at about
12,000. We can run it using up to 4 pairs. Any
recommendations on products that will reach that far? Were
There are many ways to do the screen pop, I'd like to do this way:
1. Build the manager interface as an event server, which collect agent
connet events.
2. Build a Java applet with the constant connection to the event
server, each agent starts the Java applet at first
task of each day
3. The
Check out ethernet extenders
http://www.rad-direct.com/App-Ethernet-extender-copper.htm
On Thu, 2006-07-27 at 15:39 -0600, Brian Vincent (C) wrote:
Two questions:
1. We need to run Ethernet out to a really long distance –
20,000ft. We have the ability to put a powered
Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive.
On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote:
Two questions:
We need to run Ethernet out to
If you have line of sight between the points, maybe you could setup a
wireless link point to point, I know some people who have done it
over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you
could get more).
just a thought
Joe Pukepail escribió:
Fiber? Otherwise maybe look
I would really look towards fiber, the bandwidth and distance can easily be handled.On 7/27/06, Manrique Feoli
[EMAIL PROTECTED] wrote:
If you have line of sight between the points, maybe you could setup a
wireless link point to point, I know some people who have done it
over 3 to 5 miles
Plus with fiber there's no lighting surge risk that'll burn out your equipment at both ends if the lightning hits the ground anywhere nearby.-brandonOn 7/27/06,
Bruce Reeves [EMAIL PROTECTED] wrote:
I would really look towards fiber, the bandwidth and distance can easily be handled.On 7/27/06,
Use a variable that is set when the call
comes in such as:
Exten = s,n,Set(OUTSIDECALL=1)
Then in your dial macro test for variable existence
and change ring via alert info or other distinctive ring methods. It is unfortunate
that it is heavily dependant on technology of the channel
Make sure the binary you downloaded MATCHES your machine.
snip
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- Original Message -
From: Wasif
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Thu, 27 Jul 2006 18:06:39 -0300
Subject: [asterisk-users] Getting no Audio
with G729
Hello,
Bonjour.
Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/
Hello all
I have been recently asked a question that I dont know how to possible answer
correctly.
My friend who is a tech in a small company who has a Panasonic D500 PBX super hybrid
system ( Out of dated) and one day his boss wants him to record all
conversation between the 24 lines
Hi All,
I have a problem and I'm not sure if a solution is possible without using
the asterisk testing code.
I am developing a volunteer translation service that users can dial into.
I have a list of volunteer translators cell phone numbers stored in a
mysql database along with times that they
I know.. I know fiber would be
ideal. We have single-mode all over the place. We even have some dark, unterminated
strands within 2000ft of this location it makes me want to cry.
Unfortunately lighting it up isnt an option we wouldnt
gain anything because we couldnt
Id think youd
want Asterisk to sit between the T1 and the D500. Pump the T1 directly in
and then pump it directly out. Logic in the middle to record the call is
left as an exercise for the reader.
They make off-the-shelf
products to do this. I dont know
Hi List,Does anyone know how to set up QoS on
the SNOM 360 ? Thanks.
Dovid
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Hi again,
Asterisk was not behind the NAT and I downloaded correct platform of codec.
I solved my problem by changing the prompts into G729 format. And it works
fine now.
Now I need to know about a utility which can convert all ulaw audio prompts
into g729 prompts in bulk. Or is there any was
Hi all,
Asterisk 1.4 was originally scheduled to be released early July
2006. Is there an update on the expected release of this version?
Also is there a changelog or feature list available that lists the
differences over 1.2?
TIA,
Max
--
Max Clark
http://www.clarksys.com
Hello Cavanna,,
* Cavanna, Richard [EMAIL PROTECTED] [27-07-06 15:59]:
The real thing that would help is a complete list of the configurable
comands on the latest firmware so I can create the config file.
try that config file, works perfectly for me.
Best regards,
Matthias
--
Programming
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