I have successfully configured and tested the callgroup and pickupgroup for my ZAP
interface..
but suppose I have 12 FXS interfaces, all are in a same pickup and callgroup, and
there is only a single access code to pick a call i.e. *8, and if at a time multiple
interfaces are ringing i.e.
Dear All,
I would like to install Asterisk to support my VOIP business, intending to
use Asterisk as a VOIP softswitch and/or gateways endpoints. I am
considering using either FreeBSD or Linux Redhat.
Can someone share the experience as to which OS would provide a better
environment for running
Hi,
I have Asterisk environment with the following rules:
Asterisk answer to an incoming call and Dial Operator A,
After 15 seconds if Op A doesn't answer * will dial Op B,
If even Op B will not answer * will call a group of users.
Now, what I also need but I don't know how to implement the rule
On Wed, 2004-03-17 at 04:43, Adam Hart wrote:
Eric Wieling wrote:
6) are there USA resellers
Yes, many USA resellers have expressed interest. Virbiage won't be
selling directly.
And the 255 million people in Europe? Please not the usual, 75US$ for
the unit 80US$ for FedEx or UPS to
Matteo Rancilio ha scritto:
Hi,
I have Asterisk environment with the following rules:
Asterisk answer to an incoming call and Dial Operator A,
After 15 seconds if Op A doesn't answer * will dial Op B,
If even Op B will not answer * will call a group of users.
Now, what I also need but I don't
Essentially these are general issues I have with Sipura SPA 2000:
* If SPA 2000 is behind NAT, calls are not hanged up when
receiver is replaced. I think asterisk does not get hung-up
signal from SPA so called party user agent is ringing until
timeout expires or
Dave Cotton wrote:
And the 255 million people in Europe? Please not the usual, 75US$ for
the unit 80US$ for FedEx or UPS to deliver it from the US.
No Dave, more like $125 for the unit and $80 FedEx :)
___
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[EMAIL PROTECTED]
Any volume offers allready ?
On Wednesday 17 of March 2004 12:22, randulo wrote:
Dave Cotton wrote:
And the 255 million people in Europe? Please not the usual, 75US$ for
the unit 80US$ for FedEx or UPS to deliver it from the US.
No Dave, more like $125 for the unit and $80 FedEx :)
So... Someone from Europe resell. There's a solution, no?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of randulo
Sent: Wednesday, March 17, 2004 6:22 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] The FT201 is currently being
manufactured
All I know is Virbiage seems to be very bad about distribution. I
talked with them about 4 months ago about distribution and they never
did what they said they would. They were supposed to send me paperwork
or something, never did. Now I e-mail them all the time never with a
response.
It's sad
Hello,
I've got problems to configure ip vegatsream gateway products with
Asterisk. I've got a Vega50 FXS ip gateway to connect POTS to IP phones but
when i pass a call, even if the correct port is 'responding' (the LED is on
on the vega50 interface), POTS doesn't ring...
The sip registration is
Morning folks,
two great news:
first, there's a VoIP presentation of Saarland University (with * of
course ;) at CeBIT. Have a look at http://saarland10.cebit.dfn.de/
You can find us in Hall 11 at Booth E30.
Then there's a news article on heise.de mentioning Asterisk and the
presentation our
Hi Matteo,
I have done a clean install of RH - this informs me that I have Linux
Version 2.4.20-8 - i686.
Where do I download the drivers for RH9 with capi support for our AVM C2
ISDN card?
How do I install them?
I have not yet installed * as I just want to be able to prove that the ISDN
card
Does any combination of Asterisk hardware and software
exist that supports connection to the Japanese
telephone system? (T1, E1 or J1 would be preferable,
but analog would be OK as well as a last resort (20
lines though..)).
Anyone have any thoughts or experience with this, (or
if Asterisk is
Yesterday evening, the speech on all the calls I made using VoicePulse
sounded choppy from my side, although the called party said I sounded fine.
Also, the voice mail messages I recorded calling in to the VoicePulse number
sounded choppy. Calls I made over the PSTN line using my Zap interface
Hullo!
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
Sample Scenario 1:
I call in on external line X100P. I successfully ring an
extension. The extension answers. [we have an established
call going on now] I hangup (from the external call).
Listening to the
Thanks,
this helped. As well as disabling any autodetect settings in
the IOS.
Manuel
try adding:
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind connect enable 8
to the dial-peer on the Cisco GW...
--
On Tue, Mar 16, 2004 at 05:20:28PM -0800, Andrew Gillham wrote:
Derek Bruce wrote:
depends on which Cisco softphone you are refering to... they have a few
different versions... including an NBX version which will not work with
Asterisk...
- Original Message -
From: Tim Sailer
VoicePulse has always had this problem. It's VoicePulse's poor asterisk
box.
Use NuFone. NuFone VoicePulse are actually use the same company for
voice.
We use VoicePulse as a backup and that's it
_
From: [EMAIL PROTECTED]
What sort of phone line are you using? Connecting an X100P to a PBX line
or ISDN TA can cause the problems you mention.
Iain
--On Wednesday, March 17, 2004 7:37 am -0600 [EMAIL PROTECTED] wrote:
Hullo!
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
Sample
If am trying to set up a virtual office using asterisk. I
would like the voicemail to say Unavailable (or better ..not present) when a
local extension phone is not registered and only say busy when the phone is
actually busy. I am currently using SIP for my local extensions. However
Update on this. I had the exact same issue today. At almost exactly the same time as
yesterday. Possible telco problem? Timing issue with zaptel? Never had this issue
before updating libpri as of 3/8.
Here's zaptel.conf span 7 is PRI from Verizon, span 8 is T-1 from Sprint. Dual
T400P,
For starters, you're going to want to try to post in text only format in
the future. I converted this email to plain text. Second of all, you can
search the archives here: http://asterisk.linkx.net/cgi-bin/asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Does anyone have any other information on nufone? nufone.net has got
to be the most singularly uninformative and annoying web page I've seen
for a business in quite a while...
Things like prices would be nice, also areas of coverage, etc.
Tim
--
Tim Sailer Coastal
Ahaa!
I am using a line coming out of an ISDN breakout box ..
I'll try it with a regular analog line next.
I'll let you all know what happens.
Thanks for the hint,
Willy
- Original Message Follows -
What sort of phone line are you using? Connecting an
X100P to a PBX line or ISDN TA
Try searching the forums or emailing sales, or calling sales.
Termination is ~ 2.9/minute I believe.
Toll-Free currently no monthly fee, minutes are 2.9/minute.
They only offer Michigan dids for now, plus the toll-free obviously.
Areas of coverage? Wherever you are :)
That's about it
Can anyone tell me what that strange buzzing sound is?
My system is working fine except for this problem with the X100P. After
what seems to be a random amount of time after system startup -
sometimes hours, sometimes days, the card gets bolloxed and just does
that buzzing sound when the line
Hi !
I have just installed a sound-card (AudioExcel AV512,
CMedia 8738-6ch MX). I am running REDHAT Linux
V8(linux-2.4.18-14).
When I started asterisk, I got problems related to the
sound device and rewink:
WARNING[73738]: chan_oss.c:238 sound_thread: Read
error on sound device: Resource
On Wed, Mar 17, 2004 at 09:10:42AM -0600, Matthew Marlowe wrote:
Try searching the forums or emailing sales, or calling sales.
Nah. I'll go elsewhere. Me having to search hard for information like
that tells me they're not looking for new business.
Thanks,
Tim
--
Tim Sailer
That's up to you Tim.
Good luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
Sent: Wednesday, March 17, 2004 10:30 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] NuFone?
On Wed, Mar 17, 2004 at 09:10:42AM -0600, Matthew
Nope, not nat'd. its on my internal network. I have canreinvite=no set and
still nothing.
By your response I guess you don't actually own one of these?
Mark
Steven Sokol said:
Are you natted (behind a NAT screen)?
Do you have any other SIP devices connected to the same network segment as
I have used both VoicePluse and Nufone. I have to say that the support and
the service I have gotten from NuFone is second to none. They are quick
to respond, they had me up in no time.
On Wed, 17 Mar 2004, Matthew Marlowe wrote:
Try searching the forums or emailing sales, or calling
Title: RE: VMware, * and SJphone ... newbie
I got SJphone to work and even a Cisco 7940 to connect to * inside VMware.
-Original Message-
From: Jennings, Mike
Sent: Thursday, March 04, 2004 8:42 AM
To: '[EMAIL PROTECTED]'
Subject: Re: VMware, * and SJphone ... newbie
Thanks
Is there a traceroute equivalent in the VoIP world? I would like to see the
route a call takes after it gets to the gateway. Basically showing all the
hops until it reaches it's destination or PSTN termination.
-Dave
___
Asterisk-Users mailing list
Hi !
Sorry, forgot to mention I have X100P card. So I
resend this message.
I have just installed a sound-card (AudioExcel AV512,
CMedia 8738-6ch MX) and X100P card. I am running
REDHAT Linux V8(linux-2.4.18-14).
When I started (asterisk -c), I got problems
related to the sound
hello I am not sure where to ask this question at
so please except my apologise if this is the wrong list.
I need to ask if any one has got firefly sip
version to work with fre world dialup?
if so what info did they use to
connect?
once again if this is the wrong list if the person
who is
hank smith wrote:
hello
Hi hank
I am not sure where to ask this question at so please except my
apologise if this is the wrong list.
Why not post to the fwd forum?
I need to ask if any one has got firefly sip version to work with fre world
dialup?
Not that I've heard. There is a new beta out
1) Please post in text only
2) Trying the company that makes FireFly is a good place to start,
http://www.virbiage.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank smith
Sent: Wednesday, March 17, 2004 11:13 AM
quote who=[EMAIL PROTECTED]
I have used both VoicePluse and Nufone. I have to say that the support and
the service I have gotten from NuFone is second to none. They are quick to
respond, they had me up in no time.
I have Nufone and I would have to say, their network is top. I have not had
__
Back in January I started having a problem with my Sipura (and there was
at least one other on the list with the same problem) that if I answer
an incoming call (via X100P) on line 1 of my Sipura, the caller cannot
hear any voice from the internal extension. If the internal
My inbound fax extension is set to do a goto to
ext 888, my dial line is below. I am using a triple
distinctive ring (custom) on my Zap 9 channel
exten = 888,1,Wait,2
exten = 888,2,Dial(Zap/9r3,20|d)
My setup is for 9, voice outbound calls and 8 for
outbound data calls. Portion of the outbound
Did you make the file executable. chmod 777 *, what was the message given
from konsole?
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Vikram
Rangnekar
Sent: Tuesday, March 16, 2004 2:35 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI test
On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote:
Is there a traceroute equivalent in the VoIP world? I would like to see the
route a call takes after it gets to the gateway. Basically showing all the
hops until it reaches it's destination or PSTN termination.
For SIP, there is a
If you want all of what NuFone doesn't have, then go with VoicePulse.
The fact that dialing toll-free numbers requires a completely valid and
full ANI is a well known fact.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Hajime Lanning
The search continues for a workable, affordable small fxo adapter.
While I'm still waiting to hear a real-world account of the Welltech
products, does anyone here have experience with Clipcomm's model 410
quad fxo adapter? I have inquired with them as to who resells their
products in the US.
Robert Hajime Lanning wrote:
--snipped--
Their business side (and trouble shooting) is not ready for prime time.
Issues:
o NO BILLING!
o no detailed accounting
o no way to check your account other than emailing a request.
- I may setup a cron job to request my account ballance once a
I would like to know how and where nufone and the other get their access
to provide termination, and yet only offer 2.9c /min
We are about to sign up with a termination provider and have read often
that people suggest nufone, yet for everyone who suggest them, they always
have had some sting in
[EMAIL PROTECTED] wrote:
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
snipped your scenarios
I am having the same issue on a normal analog POTS line (but in France
so you never know what other signalling anomalies there may be.)
The h signal never happens on
Can anyone verify this?
I have 2 voicemail context and when using the Directory app I seeing odd
results.
If I spesify the context as (default) I can only access default context
users as expected and it uses default extension.conf context to dial
If I specify the context as (group1) I can access
On Wed, Mar 17, 2004 at 09:47:34AM -0600, David Zuzga wrote:
Is there a traceroute equivalent in the VoIP world? I would like to see the
route a call takes after it gets to the gateway. Basically showing all the
hops until it reaches it's destination or PSTN termination.
Note that sipsak
I haven't tested this specific model of their product, but I have tested
their VOIP SIP hardphones. As of 6 months ago they had no US reseller. I
bought two test units directly from them by credit card. The work OK but
don't expect too much support or good documentation in english. Also it
would
1. broadcast intercom functions at the phone are typically associated
with key systems, and not so much on pbx's. (There are several
key system functions that don't translate nicely into pbx use.)
Rich, you will find most pabx's support all handset pages (at least all
of the fujitsu's and nec's
I have been having the same problem with 800 numbers. NuFone and
VoicePulse always behave the same (when one can't connect, neither can
the other).
I have so far found no explanation for this. Some other 800 and 877
numbers I can call.
Can you elaborate on this at all?
Thanks!
o I
For the archives
BellSouth Tariffs and Price Lists:
http://cpr.bellsouth.com/
--
Eric Wieling [EMAIL PROTECTED]
BTEL Consulting
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
I can't only for the fact that when I cant dial a number on Nufone, I
can dial it on Voicepulse.
I know the reason that I cant call the number via nufone is because the
ani is being sent wrong/invalid - But im sending it right!
Make sure your sending your caller ID with 10 numbers only, and a
I did not have intermitent access. I could not dial any tollfree number at all.
It had to do with the CallerID I was sending. It needs to be 10 digits exactly.
quote who=Matt Lawson
I have been having the same problem with 800 numbers. NuFone and
VoicePulse always behave the same (when one
Well, there ya go. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Hajime Lanning
Sent: Wednesday, March 17, 2004 1:15 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] 800 Numbers (was Re: NuFone?)
I did not have intermitent
Hi,
Seems like there arn't any alternative to NuFone either ?
Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached.
Doug
Message: 2
Date: Wed, 17 Mar 2004 08:34:25 -0800 (PST)
Subject: RE: [Asterisk-Users] NuFone?
From: Robert Hajime Lanning [EMAIL PROTECTED]
To: [EMAIL
See
http://bugs.digium.com/bug_view_page.php?bug_id=0001195
and also
http://bugs.digium.com/bug_view_page.php?bug_id=0001220
On Mar 16, 2004, at 1:18 PM, Senad Jordanovic wrote:
Miguel Cavazos wrote:
if it was related to the dsl line i would notice my other phones such
as grandstream and the
Paul Cheng wrote:
See
http://bugs.digium.com/bug_view_page.php?bug_id=0001195
This is resovled now...
and also
http://bugs.digium.com/bug_view_page.php?bug_id=0001220
This is related to above and it is solved now...
Ta
SJ
___
Asterisk-Users
I wonder if anyone could post a how-to for the chan_sccp, I've
downloaded and compiled the code, but I don't know where to go from here,
any help would be appreciated
Thanks
Robb
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
And I thought the 1 was part of a valid ANI.
It would have helped if, when I sent to Nufone that I was using these lines:
exten = _91NXXNXX,1,SetCallerID(Robert Hajime Lanning 14082729747)
exten = _91NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
That I would get the response that I
Hi Martin,
I don't think I have any playback of answers in my extension. Please,
check the following exten.conf:
[default]
exten = _X.,1,SetVar(VCOL=20)
exten = _X.,2,SetVar(VPRL=0)
exten = _X.,3,SetVar(VDIG=0${EXTEN})
exten = _X.,4,SetCIDNum(123456|a)
exten = _X.,5,Wait(1)
exten =
Hi,
I want to use my Cisco7960 with sccp. Current firmware version is 5.05,
chan_sccp Version is 0.2
I attached my sccp.conf. A lot of things doesn't work:
- speeddials are ignored, nothing is display
- Description in the top line of the display is jensp and not Jens-7960
- the first two line
Doug Harris wrote:
Hi,
Seems like there arn't any alternative to NuFone either ?
Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached.
Doug
If you want SIP/IAX termination from someone other than NuFone for the same
price, you can contact me. We can offer that.
John
Il 02:10, mercoledì 17 marzo 2004, Daniel Bichara ha scritto:
Hi All,
I have posted before asking for a Connect message sent from Zap
(ISDN/PRI - by *) when receiving a call (incoming) and dialing to
another extension. To clarify the situation, I will describe the problem:
1) My * box is
Has anyone set up a commercial IPcentrex service yet utilising the latest
SIP phones which support IAX? It seems to me that IAX is the best way
forward given the problems with NAT (and the shortage of bandwidth with BT's
ADSL services here in the UK) and that the latest phones (Verbiage) would
I hate to add to the broken record-like melange of my fax won't work
messages, but everything I've tried with all I could learn from the
archives has not yet worked to get my fax machine (an HP combo
tupperware tub) to receive a fax. In the combo's defense, I can't
verify that the incoming
Since everyone is offering their services then:
USA - £0.016 (~ 2.9c)
UK - £0.016 (~ 2.9c)
Europe - £0.02 (~ 3.6c)
UK 0800 - FREE
SIP / IAX termination. auto-provisioning, web-based billing, call
history, on-line top-up, credit-card payments.
Not US-based though :-(
Tan
www.voiptalk.org
Hi all,
I have been using Asterisk for a couple of months now with some GS
handsets and an X100P FXO card. The system works great, but I would like
to add ISDN BRI to take advantage of the extra features, faster call
setup time, etc. I was wondering if anyone could recommend any BRI cards
Price, quality, etc?
John Baker
On Wed, 2004-03-17 at 13:36, John Fraizer wrote:
Doug Harris wrote:
Hi,
Seems like there arn't any alternative to NuFone either ?
Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached.
Doug
If you want SIP/IAX termination
Your fax extension looks just like mine, except I'm using an FXS card. Mine
redirects properly, but the faxes are garbled.
I never found documentation for the d option, and it doesn't seem to help
in my case.
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
Maybe I'm wrong but you have different oprators - two different switches
and you dont synchronize with them you dont use them as your
timing source
I'd go like like this
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
span=5,0,0,esf,b8zs
Hi All,
We know everyone can offer services. May we build a interconnected *
network all over the world to offer best conditions each other? We can
set a service level agreement and try ;-)
Any one?
Daniel
[EMAIL PROTECTED] wrote:
Since everyone is offering their services then:
USA -
Hello all,
I'm thinking about getting the Plantronics DSP-400 headset for use with
Xlite softphone. I currently have a analog headset that does NOT have a
DSP on board, which gives me mediocre call quality and echo when talking to
the PSTN thru my X100P card. I have zero echo when talking thru
NOOOP!!
Unfortunately, a simple POTS line (AllTel Communications)
does not resolve the issue. It appears the problem is
somehow related to the digium card, or the drivers or what
not.
Anyone from digium monitoring this list? Is this a bug
thing?
FYI here's my zapata.conf
exten = fax,1,Dial(SIP/ata4fax) ; [1]
Faxing via SIP? Does that even work?
Faxing works for me but it is via ZAP.
I do get the messages Fax detected redirecting
to Fax extension. Which you should get irregardless
of SIP.
How are you testing this. Asterisk listens for the
Fax Tone, I see you
after unloading modul zaphfc
rmmod zaphfc
Segmentation fault
my lsmod
zaphfc 0 0 (deleted)
zaptel176864 0 [zaphfc]
all function workin but only min. time
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Wed, Mar 17, 2004 at 12:25:49PM -0800, Tor Roberts wrote:
Hi all,
I have been using Asterisk for a couple of months now with some GS
handsets and an X100P FXO card. The system works great, but I would like
to add ISDN BRI to take advantage of the extra features, faster call
setup
Jim Sneeringer wrote:
Your fax extension looks just like mine, except I'm using an FXS
card. Mine redirects properly, but the faxes are garbled.
I never found documentation for the d option, and it doesn't seem
to help in my case.
Here is how I use the d option.
exten =
Hi,
The echo problem is the X100P. The hybrid is 'unbalanced',
and basically what happens is that the outgoing sound signal
comes right-on back as an incoming signal. The reason you
don't notice it using the TDM400P is that the incoming sound
is completely 'in-sync' with you talking through the
I thought thats what http://www.iaxtel.org was all about.. :)
Daniel Bichara wrote:
Hi All,
We know everyone can offer services. May we build a interconnected *
network all over the world to offer best conditions each other? We can
set a service level agreement and try ;-)
Any one?
I have seen many postings today about the choppy sound problem. Some of
these problems were fixed with the recent change to rtp.c committed
today.
However in VoIP we usually do not have control of the quality of the
data pipe we travel over. I know there are tools that show sip proxies
traversed,
I work for a business trying to move a helpdesk to its own facility.
We're migrating a small (4 seat) call center in a few months, and *
looks like a great way to manage our IVR and queues. I've got a handle
on how * works, but know next to nothing about phone systems.
We currently get about
Oh nooo. Completely missed the boat on this one. I was thinking the exact opposite
on this. I thought that if set to 1, then the span would _provide_ timing for the
connected circuit. My span 1-6 are channel banks and I thought the 1 was providing
the timing for the banks, not the other way
Jonathan Biggs wrote:
exten = fax,1,Dial(SIP/ata4fax) ; [1]
Faxing via SIP? Does that even work?
Faxing works for me but it is via ZAP.
When I started I saw no obvious signs that it doesn't. I've seen several
references to a SIP channel in example fax exten lines, but
documentation is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
| Hi,
| The echo problem is the X100P. The hybrid is 'unbalanced',
| and basically what happens is that the outgoing sound signal
| comes right-on back as an incoming signal. The reason you
| don't notice it using the TDM400P
Ah, I was hoping to find the silver bullet, but no such luck so far. I
have tried every combination of SetCallerID and SetCIDNum in my
extensions.conf, both with and without the |a option, on both services
with no luck still.
When I call myself on our 877 number, I can see that the caller ID
Some don't do g.729 and per second billing.
These are the other things when you have to compare.
would you ?
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] NuFone?
Date: Wed, 17 Mar 2004 20:17:47 -
Organization: TelAppliant Ltd
Reply-To: [EMAIL PROTECTED]
Since
Replying to my own email here...
Bob Klepfer wrote:
Jonathan Biggs wrote:
exten = fax,1,Dial(SIP/ata4fax) ; [1]
Faxing via SIP? Does that even work?
Faxing works for me but it is via ZAP.
When I started I saw no obvious signs that it doesn't. I've seen
several references to a SIP channel
Stop beating on nufone guys..lol
Nufone is great for neetwork uptime. Support can be a big cheezy at times
but It normally dosen't seems too be too muhc of a problem. Jermey is around
most of the time too help people out if not someone else is when you call
them..Not sure whatelse
-
It's probably because you're using loop start lines, which don't offer proper
hangup detection. Switch to ground start and the X100P will work for you.
One way around the voicemail issue is to turn up the silence detection and set
a short timeout.
Search the list archives for lots more
Hi all!
I am new to asterisk. I have just installed a
sound-card (AudioExcel AV512,CMedia 8738-6ch MX) and
X100P card and compiled Asterisk. I am running REDHAT
Linux V8 (linux-2.4.18-14).
When I started (asterisk -c), I got problems
related to the sound device,rewink and chan_iax2.c:
Dave Cotton wrote:
On Wed, 2004-03-17 at 04:43, Adam Hart wrote:
Eric Wieling wrote:
6) are there USA resellers
Yes, many USA resellers have expressed interest. Virbiage won't be
selling directly.
And the 255 million people in Europe? Please not the usual, 75US$ for
the
hank smith wrote:
hello I am not sure where to ask this question at so please except my
apologise if this is the wrong list.
I need to ask if any one has got firefly sip version to work with fre
world dialup?
if so what info did they use to connect?
once again if this is the wrong list if the
http://www.tmcnet.com/tmcnet/articles/2004/031704rt.htm
Previous article by same author:
http://www.tmcnet.com/it/0104/0104PO.htm
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Bob,
Help the rest of us out now and summarize the various *.conf entries
that you have working. Might even start a new posting with a subject
that will help everyone find your samples.
Rich
Replying to my own email here...
Bob Klepfer wrote:
Jonathan Biggs
[EMAIL PROTECTED] wrote:
| Hi,
| The echo problem is the X100P. The hybrid is 'unbalanced',
| and basically what happens is that the outgoing sound signal
| comes right-on back as an incoming signal. The reason you
| don't notice it using the TDM400P is that the incoming sound
| is
I have one SIP
extension that can't logon to voicemail. The log file says
-- Incorrect password '3213' for user
'4035' (context=other)
even though the
context in voicemail.cnf says
4035 = 3213,Bill
Smith
Thanks!
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax:
Hi Everyone,
We are a service provider looking at integrating *, and
notice there are a lot of issues with the companys out there that offer
services with AIX.
If there were a $20.00 a month program which included
unified communications with the * platform or just straight
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