Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent...
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hi,
Do the following steps are required while configuring D-channel
1) In zconfig.h file of zaptel package
uncomment #define CONFIG_ZAPATA_NET
make sethdlc-new
make install
2) modprobe wcte12xp
ztcfg
3) sethdlc hdlc0 cisco
Step 3 is
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Yes I do. I even did a pri debug span 1 and when I call the asterisk
box, it sees nothing.
On 7/26/07, Idris AVCI [EMAIL PROTECTED] wrote:
Do you have any extension in default context of your extensions.conf
file to accept incoming calls ?
It must be something like;
exten =
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email
All these repeated list replies with Autoreply: Autoreply: Autoreply:
Autoreply:... subjects are irritating at best and debilitating at worst!
This makes the list waste bandwidth and my inbox (and the archives too)
unreadable!
Thx!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hi guys,
I don't know if this is the question ...
But I have my softphones set up to use ilbc (because I found that it is better
for me) and I'm trying to connect them to my provider that provides me
termination through g.729. I don't have any T1 or whatever card in my server.
But everytime I
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email
hi , i think everybody is receiving theses mails from rp.
can someone unsubscribe or do something , its really annoying
thanks
--
Cheikhou DIAW
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
On Fri, Jul 27, 2007 at 10:17:08AM +0800, GNUbie wrote:
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This
[Something seems to have went wrong with my previous
posting. It appears on the archive page in another thread. I
did not receive anything myself. So I may give it another
try:]
Hi,
the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
Hello!
I'm using Asterisk 1.4 with Dialogic Diva Server Analog 8P (with CAPI) and I
need
to find a way (it can be tricky) to get the DIAL STATUS of the call when I
use
the auto dialout queue. I know the DIALSTATUS variable can only be used with
Dial
application, but I have to make difference
You can listen or join the Asterisk Users Conference Fridays at 12:30 PM
EDT
Today's subject suggestions:
FAX capabilities, what's your solution?
Multiple asterisk server implimentation: ENUM, DUNDI or even two servers
connected
Your subjects?
Share your ideas, ask your questions!
See
[Something seems to have went wrong with my previous
posting. It appears on the archive page in another thread. I
did not receive anything myself. So I may give it another
try:]
Hi,
the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
[Something seems to have went wrong with my previous
posting. It appears on the archive page in another thread. I
did not receive anything myself. So I may give it another
try:]
Hi,
the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
hi asterisk users
How can i make asterisk ignore invalid extensions, and go on playing the
background soundfile?
Normally, asteriks will take the user to the invalid extension if the caller
presses anything other than 1 or 2 in the following context::
[example]
exten = s,1,Answer()
exten
Try with this (it could save your time):
http://www.queuemetrics.com/
http://www.asteriskguru.com/tools/queue_stats.php
http://www.bicomsystems.com/home/C/P/731/143_3604/
Kind Regards,
Faruk.
[EMAIL PROTECTED] wrote:
Greetings, list!
My boss would like some statistics on how many calls are
On Fri, 2007-07-27 at 06:46 -0700, bilal ghayyad wrote:
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
These are global variables as defined in the [global] section of
extensions.conf. They're simply variables that can be used later on in
your dialplan.
I
You can listen or join the Asterisk Users Conference Fridays at 12:30 PM
EDT
Today's subject suggestions:
FAX capabilities, what's your solution?
Multiple asterisk server implimentation: ENUM, DUNDI or even two servers
connected
Your subjects?
Share your ideas, ask your questions!
Hi List;
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
I know SIP/John and Zap/1 but I do not know above (I
do not know also the difference between Zap/2 and
Zap/g2)?
Any advise?
Regards
Bilal
Dear all
I have asterisk and now i want to configure meetme confranceing
but problem is when i dial confrance number i got message conf number is
invalide and i got this error /dev/zap/psudo device not found what is this ???
-
Need a
On Fri, 2007-07-27 at 06:26 -0700, bilal ghayyad wrote:
For example: what is the best (shortest) way to search
for information related to the command playbak()?
I find that the fastest and most up-to-date information regarding the
dialplan applications is the online help in the Asterisk CLI.
bilal ghayyad wrote:
Hi List;
I am trying to use wiki via the link
(http://www.voip-info.org/wiki/index.php?page=Asterisk)
in effective way to find the needed resource for me,
but still it is hard to arrive for the needed
information.
For example: what is the best (shortest) way to
Using the CLI is another good way to find that information quickly:
nox*CLI core show application Playback
-= Info about application 'Playback' =-
[Synopsis]
Play a file
[Description]
Playback(filename[filename2...][|option]): Plays back given
filenames (do not put
extension). Options may
Hi List;
I am trying to use wiki via the link
(http://www.voip-info.org/wiki/index.php?page=Asterisk)
in effective way to find the needed resource for me,
but still it is hard to arrive for the needed
information.
For example: what is the best (shortest) way to search
for information related to
Who's definition of advanced are you going to use?
I suggest a way forward might be...
1. List all the features of Asterisk.
2. Cross out the features you consider non-advanced.
3. Whatever is left is what you wish to study.
:)
From: [EMAIL
James FitzGibbon wrote:
On 7/26/07, *James FitzGibbon* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Is it possible for qe.parent-membercount to be set to zero in a
queue where all agents but one are on the phone and that one
remaining agent lets their phone ring without
Hello,
I am interested in knowing what are the advanced topics that can be learned in
Asterisk. It would be helpful if there are any reference books or tutorials on
Asterisk that cover advanced concepts on Asterisk.
Thanks in advance!
A Successful Person Is The One Who Can Lay A Firm
On 7/27/07, James FitzGibbon [EMAIL PROTECTED] wrote:
I'll go open a bug report.
http://bugs.digium.com/view.php?id=10320
For anyone who wants to track it.
--
j.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On 7/26/07, James FitzGibbon [EMAIL PROTECTED] wrote:
Is it possible for qe.parent-membercount to be set to zero in a queue
where all agents but one are on the phone and that one remaining agent lets
their phone ring without answering it?
I added some debug code to app_queue and ran a few
Considering getting the card pictured here,
http://www.voipsupply.com/product_info.php?products_id=2419searchid=341962,
and was wondering if anyone had any experience with those in a 2950's
case. From the picture it looks to be a pretty large card. I'd hate
to get it and have it not fit!
--
On Fri, 2007-07-27 at 05:21 -0700, satish patel wrote:
I have asterisk and now i want to configure meetme
confranceing but problem is when i dial confrance number i got message
conf number is invalide and i got this error /dev/zap/psudo device not
found what is this ???
The
Hi,
I have a Digium B410P 4-port BRI card.
I installed misdn 1.1.3 with hfcmulti driver and
misdnuser 1.1.3.
I configured the card correctly as misdnportinfo
reports:
# misdnportinfo
Port 1: TE-mode BRI S/T interface line (for phone
lines)
- Interface is Poin-To-Point.
- Protocol: DSS1
On 7/27/07, Jared Smith wrote:
[...]
As a second suggestion, I'll put in a shameless plug for the many
different Asterisk books on the market. (Yes, I admit it, I helped
author one of them.) Many of them have reference sections that are
indexed and list the dialplan applications
Can someone comfirm my logic here?
If I want a phone to use G729 I can set it to use G729... do I
also need to set it in Asterisk? I'm thinking no... as long as
asterisk WILL do G729... if that's all the device accepts it should go
to that codec, yes?
Ciao Baji,
[...]
As a second suggestion, I'll put in a shameless plug for the many
different Asterisk books on the market. (Yes, I admit it, I helped
author one of them.) Many of them have reference sections that are
indexed and list the dialplan applications alphabetically.
I
On Fri, 2007-07-27 at 11:09 -0500, Victor Toofic wrote:
Hi,
zaptel.conf:
span=1,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
loadzone=mx
defaultzone=mx
unicall.conf
[channels]
context=incoming
usecallerid=yes
hidecallerid=no
Hi,
I've recently upgraded Asterisk to the latest version 1.4.9
on a PBX that manages several queues, but at least on one queue strategy
(leastrecent) it doesn't seem to be distributing the calls has it should.
I think this strategy should work like
Yes it seems... but not just with that strategy. Now I'm working on
a queue witch I need to go to agent 1004, then 1018 and then 1010 so I wrote
this in queue.conf:
[general]
persistentmembers = yes
autofill = yes
monitor-type = MixMonitor
[FAC]
musicclass = default
strategy =
Hi all,
I'm new to the list, so I apologize in advance if I'm beating a dead horse
by asking this, but I read somewhere that asterisk 1.4 has MWI working for
Avaya and their rather troublesome SIP firmware. Can anyone verify this
before I go changing phone systems around?
Thanks
Derek
Dnia 2007-07-27, o godz. 11:09:37
Marco Campos [EMAIL PROTECTED] napisał(a):
I think this strategy should work like this:
a) Make a list of available agents and their idle time (time
since last call) and update it each time a call gets answered or ends
b) When a
--- Vieri [EMAIL PROTECTED] wrote:
Unable to load module chan_misdn.so
Never mind. I found out it was a permissions problem.
Either asterisk must be run as root or the asterisk
user must be part of the dialout group.
(tweaked /usr/sbin/amportal)
Hi,
I've finally got running Asterisk 1.2.14 with UniCall MFC/R2 patches, I
can generate calls and all seems OK but I cannot receive any call, this is what
I get:
MFC/R2 UniCall/3 - 0001 [1/IDLE/Idle /Idle]
MFC/R2 UniCall/3 Detected
MFC/R2 UniCall/3 Creating a new call
On 7/27/07, Mark Michelson [EMAIL PROTECTED] wrote:
Could you submit this as an issue on the bugtracker? The 'n' option was
mucked with just prior to the 1.4.9 release and so any problems
experienced with it should be reported there so they can be fixed as
quickly as possible.
It's been
http://www.voip-info.org/wiki-Asterisk+cmd+Playback
you can use google asterisk cmd playback..
bilal ghayyad wrote:
Hi List;
I am trying to use wiki via the link
(http://www.voip-info.org/wiki/index.php?page=Asterisk)
in effective way to find the needed resource for me,
but
Over the last week we've been having issues on our Telco provided TDM T1
with the circuit bouncing for several seconds and restoring itself back
into service. The T1 is using a TE410P. On the CLI, I see message that
span 1 is yellow alarm, then restoring.
I reported this problem to the phone
On 7/27/07, Matt [EMAIL PROTECTED] wrote:
Can someone comfirm my logic here?
If I want a phone to use G729 I can set it to use G729... do I
also need to set it in Asterisk? I'm thinking no... as long as
asterisk WILL do G729... if that's all the device accepts it should go
to that
To give everyone our motives or circumstance:
The company in question does not pay for the salesperson(s) mobile phone
therefore, we have a mixed bag of phones and providers for them. None are
smartphones, none sync any type of e-mail.
Our company currently pays for alpha-numeric pagers
Personally, I always find what I need with:
1. www.google.com
2. site:voip-info.org cmd Playback
3. Search
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
On 7/27/07, Andrea Spadaccini wrote:
Ciao Baji,
[...]
As a second suggestion, I'll put in a shameless plug for the many
different Asterisk books on the market. (Yes, I admit it, I helped
author one of them.) Many of them have reference sections that are
indexed and list the
El Fri, Jul 27 de 2007 a las 11:18 -0500, Carlos Chavez comentaba:
Try changing protocol variant to: protocolvariant=mx,10,4
I've already tried several combinations. I dont know exactly what means
each number but I could get an idea according what I saw in the logs.
Even so, I've tried it
Kyle Sexton wrote:
Considering getting the card pictured here,
http://www.voipsupply.com/product_info.php?products_id=2419searchid=341962,
and was wondering if anyone had any experience with those in a 2950's
case. From the picture it looks to be a pretty large card. I'd hate
to get it and
I'm running Asterisk without FreePBX or any of the other managers. I'm
trying to figure out how to set the maximum number of channels allowed on a
single line? I'd just rather not have Asterisk try the line when I know
I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this
baji, mhoppes,
remember, if you have Only the g729 codec allowed or if this is the
only allow= entry in the sip.conf file, callers requesting any other
codec will be rejected
daveC
Baji Panchumarti wrote:
On 7/27/07, Matt [EMAIL PROTECTED] wrote:
Can someone comfirm my logic
Anybody here having any problems with nufone?
Calls are not going thru, when trying to call their customer service
number it doesn't go thru.
When trying to resolve www.nufone.net I get (sourec:
http://www.dnsstuff.com/tools/lookup.ch?name=nufone.nettype=A ):
How I am searching:
Searching for
randulo,
I could not get into the conference today... the SIP line was busy, no
matter what I do, the website thinks I'm not logged in and gives me the
login page. after I login, anything I want to do brings me back to the
login page... so I tried to re-setup the account thinking I wasn't
Quoting C F [EMAIL PROTECTED]:
Anybody here having any problems with nufone?
Calls are not going thru, when trying to call their customer service
number it doesn't go thru.
When trying to resolve www.nufone.net I get (sourec:
http://www.dnsstuff.com/tools/lookup.ch?name=nufone.nettype=A ):
I'll assume you mean a Dell PowerEdge 2950. Sangoma's web
site says the
cards dimensions are 55mm(H) x 290mm(L). A Full-Length PCI card is
107mm(H) x 312mm(L). According to the PowerEdge 2950 Getting Started
Guide Page 10:
Left riser
PCI-X option: two full-height, full-length
Right.. what I'm asking is:
If I set my PAP2T to use G723 or G729 outgoing calls from that
device go in that format.
However, incoming calls to the device from asterisk are running at
G711u. The PBX will access any format G711u, G723, G729 or GSM.
What do I need to do to make asterisk use
Noah,
Thanks for the input. I'm thinking the problem with the stop
gracefully is that it would confuse the auto fail-over appliance, in
that it would either detect the server is dead and hard switch the
T1s or keep sending calls there which Asterisk would reject.
I'm thinking a better
in ur sip.conf under the device definition you can set it
for example device name is asterisk is pap2
[pap2]
username=pap2
secret=blabla
type=friend
disallow=all
allow=g729
Then asterisk will only use g729 for incoming as well as outgoing calls on
this device .
On 27/07/07, Matt [EMAIL
But why dont they have *any* phone lines for support? Or at least a
busy signal? Why is their DNS failing?
On 7/27/07, Shane Young [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
Anybody here having any problems with nufone?
Calls are not going thru, when trying to call their
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
*
call-limithttp://www.voip-info.org/wiki/edit.php?page=Asterisk+sip+call-limit
* = number : Number of simultaneous calls through this user/peer
On 27/07/07, Nicholas Blasgen [EMAIL PROTECTED] wrote:
I'm running Asterisk without FreePBX
quote who=C F
Why is their DNS failing?
Looks like ns1 is down. Probably their master DNS server.
ns2 is up, but looks like their zone expired, since it could not refresh
from ns1, so it is no longer reporting authoritative for nufone.net.
They should look into longer expiry times on their SOA
On Wed, 2007-07-25 at 13:02 -0500, Eric ManxPower Wieling wrote:
Short Answer: No.
Long Answer: Maybe. If you can get your device to send inband DTMF and
tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should
just pass the DTMF as audio. Then if the call goes via IAX2 it
quote who=C F
Why is their DNS failing?
Looks like ns1 is down. Probably their master DNS server.
ns2 is up, but looks like their zone expired, since it could not refresh
from ns1, so it is no longer reporting authoritative for nufone.net.
They should look into longer expiry times on
Ok, my boss is telling me that Im using Category 1 in the signaling and he
is asking me to change it to Category 2.
R2 Incoming Voice(0/0): DSX (E1 0/0:0): STATE: R2_IN_CATEGORY R2 Got Event
R2_TONE_OFF
Enter r2_comp_category
r2_reg_generate_digits(0/0:1(1)): Tx digit '#'
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