Anybody know how to match under sip.conf and cisco 53xx ? It looks like due
to dynamic port number, it is not able to authorize it.
Here is what I get under debug
Using latest request as basis request
Sending to 216.236.160.15 : 5060 (non-NAT)
Found no matching peer or user for
I got some advice from Josh Colp that has helped with some of my problem:
it may have a little logic flaw in the way transcoding is supposed to be done, from
the way your message is I would say you are getting hit by this. (Upgrading to latest
CVS head will fix it) but one solution is to be the
Hi Charl,
Im sorry to tell you a disappointing comment about the PA168 IP Phone, I
have here too such like that and it's a crap... It works in a while,
sometimes it can even send eight digits ( I mean it fail after dialing
around 4 to 5 digits).
-Original Message-
From: [EMAIL
Hi Basher,
Currently im using my A800 Quintum registered in my Asterisk SIP server. For
you to register your Quitum to Aterisk, define your asterisk in as proxy and
registrar IP at SIP config at quintum (I can give you the sample config at
private mail). Also setup a user account at
Yeah the idea: its like a karaoke conversation between people via
voicemail thats posted on a website as an audio thread under a creative
commons licence.
Base requirements:
-record: pick up, and reply to recordings.
--listening to music while recording
--having that music mixed with the
Hi,
what do i have to configure to get a busy tone when dialing out over
ISDN channel with my Polycom 500 IP?
Kib
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-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Jean-Michel Hiver
Envoyé : lundi 25 avril 2005 07:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] NEED HELP PROGRAMING ASTERISK
VoIP NETWORK
Franz
i am using X100P on RHEL4, all incoming calls doing
well, during any outbound call from sip to pstn, it
hangup right away when the remote side pick up the
phone.
i've been trying to trace out this problem for 2days.
for the log snapshot below,
DEBUG[2401]: Exception on 15, channel 1
DEBUG[2401]:
Marc Storck [EMAIL PROTECTED] writes in reply to my question:
add
canreinvite=no
to the sip user definition blocks for the SIP provider and for the SIP ATA.
Regards,
Unfortunately, I already have this parameter in the sip user definitons,
as well as
a t option in the Dial command, both of
I am using Asterisk with SIP phones.
is it possible to press a key during a conversation and get
asterisk to do something? Like the # key, but I would like asterisk to
take other actions instead of transfering.
tulika
_
Your [EMAIL
Description
Amatix will instantly transform your computer in a
small PBX. You don't have to install any software,
just plug the Amatix CD in your CD drive and let the
computer boot from it. In few minutes you will get a
running Linux system with a configured Asterisk PBX.
Highlights
* Amatix
[EMAIL PROTECTED] wrote:
On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote:
Using most recent CVS-HEAD and my terminal keeps changing colors.
I'm using vt100 terminal emulation. How can I turn off asterisk's
colors? Or at least turn off the black background. My normal
terminal is
Hi
Iam new to asteriks, i juz installed it on my system and also got hold of diax(on a windows client) to call the asteriks server,now before buying any diguim hardware i want to test asteriks by making both the computers talk. I dont have any kind of h/w now, i need help from u guys to make
You know, that's exactly what I was looking for since the beginning!
Unfortunately I only found one of these items for sale in the US and even then
I'm not sure if it will be compatible with the European system! Maybe someone
can enlighten me once and for all as far as the differences between
Hi
Can you please do advertising for your company in Asterisk-Biz
Sorry, I did use the 'reply to sender' functionality but this mailing
list is utterly broken because it replaces the reply-to with the list
address.
I was just replying to the poster. I am very sorry for the inconvenience.
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Another stupid question now: anyone knows who does the voices in all
these nice systems ? Like, Welcome to Mycompany, for sales press 1,
for support press 2
Allison Smith. See http://www.digium.com/index.php?menu=thevoice
and
Another common problem that causes echo in networks is not setting your
loss plan correctly. You need to be sure that you aren't coming in too
hot at any of your analog interfaces. In general you should see a signal
between -20dbm and -12dbm when someone is talking on the line. If it is
Hi !
I am trying to configure a E100P card with Channel bank,
but I am a bit confused witheach 64K channel's ABCD bits.
The * will be connected to a PSTN switch with E1 Channel
Bank lines. The E1 lines will be used for incoming calls as FXS
channels.
My problem now is where to find the
Title: Message
Hi
all,
After many
complaints (including car manufacturers saying the american prompts are
unexceptable, EEEK) I started on a quest for real "English" asterisk
prompts.
The only one I have
found is here http://www.g7ltt.com/VoIP/vmfiles.html
And no nothing else on the
Ooops damn Outlook to Hades.
Forgot to format in plain text.
If you have been offended by this please feel free to ignore this
thread.
If not then I have left the original message below (this isnt a top post
I swear)
Thanks again
alex
-Original Message-
From: Alex Barnes
Sent: 25
On April 24, 2005 11:58 pm, Lee Howard wrote:
Certainly I can understand that Digium doesn't stand to make much money
selling X100Ps at $10 each, and I can certainly understand them choosing
to not sell them. But, by the same token I cannot understand the
community's interest in discouraging
On April 25, 2005 12:25 am, Kerry Garrison wrote:
What year is this? 2005 right? Doesn't everyone on the planet know that you
get what you pay for these days? If you want to experiment with Asterisk
there is nothing wrong with using clone X100P cards at $6.95 a pop. If you
No there is
On Thu, 2005-04-21 at 17:52 -0400, Matt Roth wrote:
Daniel,
You're correct that if we instructed the Monitor command to mix the
files the mixing would occur on the master server. I looked at the
documentation and source (res_monitor.c) of the Monitor command to
confirm that the default
Alex
I too am on the hunt for the same. I am hoping that my good friend
with the recording studio and his lovely wife will be able to perform
this.
My only issue at the moment is getting the scripts that was worked to,
failing that, next weekend I am spending hours writing down what
alison says
Joseph Gutowski [EMAIL PROTECTED] wrote:
[...]
I wasn't suggesting Asterisk should magically be able to pick up the
call before it rings at all, just that if my old roommate could
manage to dive across the room and pick up half way through the
first ring 99% of the time, surely a computer
On 4/22/05, Stefan Gofferje [EMAIL PROTECTED] wrote:
Hi folks,
I'm using a Fritz!PCI with chan_capi 0.3.5.
I found that chan_capi neither seems to signal Busy or Congestion to
callers from ISDN nor does it seem to set HANGUPCAUSE, CAUSECODE or
DIALSTATUS if an outgoing call fails. There is
Hi,
I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The
problem now comes in the PCI ports. Is there any PC that can handle 16 ports?
What is most optimal solution?
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If i'm understanding this correctly, you shouldn't need 16 ports.
If you buy 2 TDM400P cards, and load them up with 8 FXS (4 on each card)
then buy 2 TDM400P cards, and load them up with 8 FXO (4 on each card)
This should reduce your PCI count down to a more manageable 4 cards
In total your
The Polycoms also include a power supply and SIP firmware, which the
Ciscos
do not. Overall I just think the Polycoms are a better value.
Cisco SIP firmware does not support subcribe/notify method (busy
line/extension status).
Polycom does support it.
I have cisco phones right now and I'm
Make sure you have canreinvite=no in your sip peers definition, and/or
that you pass 't' or 'T', to the Dial statement.
Julian J. M.
On 4/25/05, Tim Pushor [EMAIL PROTECTED] wrote:
Hi all,
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple
[EMAIL PROTECTED] wrote:
Alex
I too am on the hunt for the same. I am hoping that my good friend
with the recording studio and his lovely wife will be able to perform
this.
My only issue at the moment is getting the scripts that was worked to,
failing that, next weekend I am spending
On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote:
I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The
problem now comes in the PCI ports. Is there any PC that can handle 16
ports?
What is most optimal solution?
The most optimal solution would be a TE110P + a channel
Thanks very much for this info Andrew.
Selon Andrew Kohlsmith [EMAIL PROTECTED]:
On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote:
I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The
problem now comes in the PCI ports. Is there any PC that can handle 16
ports?
Interestingly enough I'm looking to do the same for a Canadian English
version... does anyone to collaborate on this one?
Ian Pattison, Senior Analyst
Technology Associates Inc.
Tel: 905-459-2100 ext. 204
Mobile: 416-568-6548
E-mail: [EMAIL PROTECTED]
WWW: http://www.technologyassociates.ca
Another common problem that causes echo in networks is not setting your
loss plan correctly. You need to be sure that you aren't coming in too
hot at any of your analog interfaces. In general you should see a signal
between -20dbm and -12dbm when someone is talking on the line. If it is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Mon, 25 Apr 2005, David John Walsh wrote:
Alex
I too am on the hunt for the same. I am hoping that my good friend
with the recording studio and his lovely wife will be able to perform
this.
My only issue at the moment is getting the scripts that
-Original Message-
From: Ron Wellsted [mailto:[EMAIL PROTECTED]
Sent: 25 April 2005 13:48
To: David John Walsh; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK (english) sound files
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On
Ian
you do realise that alison is actually canadian :)
(well as far as I know she is)
On 4/25/05, Ian Pattison [EMAIL PROTECTED] wrote:
Interestingly enough I'm looking to do the same for a Canadian English
version... does anyone to collaborate on this one?
Ian Pattison, Senior Analyst
Hi there
I am looking for an open-source softphone / control for
windows that I can use in a VB 6 application that will be for commercial use. I
also need support for GSM, ulaw / alaw and possibly ilbc / speex.
I have found a couple of possibilities, but none of them
quite suit my
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Monday, April 25, 2005 8:04 AM
snip
Channel banks are great; the better ones (Adit600) can do far-end
disconnect
supervision and I think pretty much all of them do dynamic impedance
I haven't played with it yet but from the info I read I understand that I
can specify to record conversations on any extensions with the Monitor
command. Is there any interface for replaying these recordings? I'm thinking
of something like the CDR analysis package asterisk-stats with a link to the
Rich Adamson wrote:
One way is to buy a relatively inexpensive analog transmission test
set ($400 US). Most have a tone generator and level meter built in.
You didn't mention which country you're located in, but ensure whatever
test set you purchase, that it supports the line impedance in use by
[EMAIL PROTECTED] wrote:
Ian
you do realise that alison is actually canadian :)
yeah.. she is as far as I know as well...
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To
I'm nothing understand now. I have Cisco ATA 186 with one analog phone and
the following problem:
The next config works just fine:
sip.conf:
[150]
type=friend
port=5060
context=officepbx-outgoing
qualify=yes
secret=password
user=150
username=150
fromuser=150
defaultip=XXX.XXX.XXX.XXX
On April 25, 2005 09:05 am, Karl H. Putz wrote:
What configuration do you need to do to the Adit in order to get it to
recognize
FXO side disconnect? I have tried a number of different settings and can
never
get it to pass through to *.
It just worked for me, nothing unusual or funny. You
The Triplett Model 4 is one model. There are many others.
Thanks, I found something on Ebay for $120 - great advice, I appreciate it.
Chris Mason
www.anguillaguide.com
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On April 25, 2005 09:09 am, Steve Underwood wrote:
What's wrong with a little software for * to do all of the above and far
more without cost or inconvenience?
Do we not already have this with ztmonitor and app_milliwatt.so? That's what
I used, at least on Zap interfaces. If you need to
Hi
folks,
I've been trying to
get Identapop Pro to work properly and am having no success for inbound Caller
ID and Name. I've upgraded to the recent release of
asttapi.
Calling from
Microsoft Outlook contacts works great.
However on any
inbound call Identapop Pro reports the following:
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote:
Using most recent CVS-HEAD and my terminal keeps changing colors.
I'm using vt100 terminal emulation. How can I turn off asterisk's
colors? Or at least turn off the black background. My
Hi,
I am in the process of setting up an Asterisk-based PBX at work. I get
the concept of how Asterisk works pretty decently. I am more confused
about the proliferation of TLAs like FXO, FXS, TDP, SIP,
After some intense reading I have come to some understanding of the
hardware I need to
One way is to buy a relatively inexpensive analog transmission test
set ($400 US). Most have a tone generator and level meter built in.
You didn't mention which country you're located in, but ensure whatever
test set you purchase, that it supports the line impedance in use by
your telco.
The Triplett Model 4 is one model. There are many others.
Thanks, I found something on Ebay for $120 - great advice, I appreciate it.
I seen that one too. Be careful... it doesn't imply the meter is in
working order, nor any warranty, etc.
___
Thank you!
Did you change the default transfer key? Doesn't the sipura 'eat' the #'s?
yes, at least I know it should work (as I suspected).
Thanks again,
Tim
David John Walsh wrote:
call parking and transfer works great for me, on a variety of devices noteably:
sipura 2000 / 3000
xten x-lite
snom
You need a V92 capable modem for your client and a V92 capable access
server for you. The feature is called modem on hold, it lets you
pick up a call without loosing your internet connection, and resume
the dialup session after hangup. The only feature you need for your
telco is call waiting. It
Hi All,
Still having problems :-(
I have an Asterisk 1-0-7 setup on Debian 3.1 (Sparc)
I have severel SIP phones that call between each other and can chat no
probs. I can even call from the SIP phones to the sccp 7920 no
probs
However when I call from the 7290 to any SIP phone
On Sun, 2005-04-24 at 11:15 -0700, Thomas Miller wrote:
For example, Australia phone numbers can be either 6
or 7 digits, while USA phone numbers are always 10
digits.
No, they aren't Most 'local' phone numbers are 8 digits, long
distance (ie, including area code) they are 10 digits. That
It really depends on how you want the whole thing to function. After the
call is done you could simply use system to email the recording as an
attachment or you could use a php page to list all of the recordings.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The box isn't doing anything else at all. It just started this problem
recently, the only change I can correlate it to is moving to Asterisk
1.0.7.
I'm pretty much not able to use the TDM card anymore now. I am
thinking of just offering it upon ebay; there doesn't seem to be
anything I can
Message: 13
Date: Mon, 25 Apr 2005 09:08:15 -0400
From: Chris Mason (Lists) [EMAIL PROTECTED]
Subject: [Asterisk-Users] Call Recording via monitor
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type:
Dan Perik wrote:
Michael Lyszczek wrote:
I have broadvoice and they suck lately.
Can you elaborate?
- Dan
Yes, please elaborate. Do you mean to say they didn't suck previously
but now they do suck? I can't imagine them staying in business much
longer if that is truly the case.
On April 25, 2005 10:45 am, Trent Tuggle wrote:
The box isn't doing anything else at all. It just started this problem
recently, the only change I can correlate it to is moving to Asterisk
1.0.7.
So go back to the version that was working... I'm trying to help you figure
out what
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something with your *
box. This is quite different from most * sites around, that are either
questions-and-answers forums or are dedicated to documenting a feature.
Hi Michael and Tony
I have the same problem here and I have been able to check that this
problem can be solved disabling VAD in h323 destination routers, I
think this is a common problem with h323 and oh323 modules users and
for me has become a nightmare because my service provider can no
longer
Our experience with BroadVoice over the past two months:
Pros
Good voice quality
Zero downtime (not counting our ISP going down several times)
Solid connections
Low ping times
Cons
Would be nice if they supported more codecs (nothing new there)
Takes on average of 45 minutes to talk to tech
Hi -
We've been using IAX forwards between sites for a little while now
(with centralized VM). For the most part, it is fine, but I have some
very minor, yet persistent QoS issues on calls over the IAX forwards.
For most normal calls, there are very occasional minor glitches, just
an
lenz [EMAIL PROTECTED] writes:
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something with your *
box. This is quite different from most * sites around, that are either
questions-and-answers forums or are
In article [EMAIL PROTECTED],
Rafael J. Risco G.V. [EMAIL PROTECTED] wrote:
Hi Michael and Tony
I have the same problem here and I have been able to check that this
problem can be solved disabling VAD in h323 destination routers, I
think this is a common problem with h323 and oh323 modules
In data Mon, 25 Apr 2005 17:33:14 +0200, Bruno Hertz [EMAIL PROTECTED] ha
scritto:
Good idea, but don't we have already the Wiki tips/hints, editable by
anybody ? I understand people like to contribute, which is great. But
spreading the info all over the web instead of centralizing it might
be
Hello all.
I'm trying to see if anyone knows of an alternative solution, commercial
or non-commercial, to SpanDSP. I'm specifically looking for another
software-based, DSP fax that doesn't require me to add a tie up a bunch
of extensions on my PBX.
Has anyone ever seen such an animal, or gotten
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED] wrote:
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something with your *
box. This is quite different from most * sites around, that are either
In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield
[EMAIL PROTECTED] ha scritto:
In article [EMAIL PROTECTED], lenz [EMAIL PROTECTED]
wrote:
Hello,
if anyone is interested, there is a new wiki about Asterisk recipes,
i.e. step-by-step descriptions on how to perform something
I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3 machine. I
also set up the system to use Realtime for the voicemail mailboxes. I am
successfully using Realtime for extensions and sip clients on this machine,
but as yet, cannot get the voicemail system to recognize the mailboxes as
Anobody had any problem with GS ata 286? The past few days Ive been having
some problem with it, while making a call or during a call, I suddely hear a
low noise like a car engine starting and then the ata dies, as if it got
stuck or frozen.
Anybody had these problems?
your queue recipie,
does that monitor record from when the agent answers or the music on
hold prior to taking the call?
thanks
On 4/25/05, lenz [EMAIL PROTECTED] wrote:
In data Mon, 25 Apr 2005 16:12:08 + (UTC), Tony Mountifield
[EMAIL PROTECTED] ha scritto:
In article [EMAIL
Michael Welter wrote:
Mark Johnson wrote:
I tested and I do in fact get from 40-50% system util every 5 seconds
or so. After removing the wctdm module, the system util drops to 0
and stays there. I have not loaded the wcfxs and wcfxo modules
because I could never get them to work right. I
Interesting.
Can anyone out there tell me how many concurrent Monitors an Asterisk
box can handle under my scenario (see below)?
1) Monitor commands are executed on the Asterisk server.
2) Audio packets are saved to files on a remote machine via mounted drive.
3) All handling of the audio files
I got the same problem with 04/19/05 CVS version. I am using
Grandstream phones. I also noticed that when this happens, an already
hung-up call was still shown as bridged between a SIP phone and a Zap
channel.
On 4/18/05, Shaun Tierney [EMAIL PROTECTED] wrote:
I am currently running
Edwin Horton wrote:
I have the latest Asterisk CVS as of 4/21/05 and a Fedora FC3
machine. I also set up the system to use Realtime for the voicemail
mailboxes. I am successfully using Realtime for extensions and sip
clients on this machine, but as yet, cannot get the voicemail system
to
I wasn't aware that SpanDSP tied up a bunch of extensions.
Jeremy Melanson wrote:
I'm trying to see if anyone knows of an alternative solution, commercial
or non-commercial, to SpanDSP. I'm specifically looking for another
software-based, DSP fax that doesn't require me to add a tie up a bunch
Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.
I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?
Am I alone in this?
Naturally from when the agent and the user start talking to each other.
l.
In data Mon, 25 Apr 2005 17:45:00 +0100, David John Walsh
[EMAIL PROTECTED] ha scritto:
your queue recipie,
does that monitor record from when the agent answers or the music on
hold prior to taking the call?
thanks
On
Tim Pushor wrote:
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple month old version of asterisk from
CVS HEAD.
Before I go ripping things apart, I was really wondering if this is by
design, or should it work on all my devices? I have an
I am having the same broadvoice issue at the moment.
jerry
Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it cannot
complete the call.
I can make outgoing calls from my system through broadvoice
I modified the source code as I have 10 msn numbers here at home, I will try to
make a diff of the changes.
Peter
-Original Message-
From: Stefan Helbing [mailto:[EMAIL PROTECTED]
Sent: 22 April 2005 16:40
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] chan capi:
Kib Eki wrote:
Hi,
what do i have to configure to get a busy tone when dialing out over
ISDN channel with my Polycom 500 IP?
Try priindication = inband in /etc/asterisk/zapata.con
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Does Mark post on this list?
Hmm. Let me think about this one ;-}
I was trying to get a movement going to make sound files in other
English language variants but it all seemed to die off.
I have not found the time to complete the Southern England Male prompts
but if you send me a list of the
I seem to be down right now too.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Max Clark
Sent: Monday, April 25, 2005 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Broadvoice Down?
Is anyone
Hi Everybody can someone tell me why I can hear audio? My call is to my
proxie which is directing it to my Asterisk box. The Voice mail is
playing but I think its playing to my proxie.
the phone is on 198.31.185.246:63257
Here is from the sip debug. Thanks
Sip read:
INVITE sip:[EMAIL
Yes, no problem.
Seems like asterisk is not 'breaking in'. I have it in the media path
and dtmf all works properly. The sipura devices are a mismatch of
codecs, but the 841 is g729 and the iaxy is of course ulaw. I am running
the IPP g729 codec and I did wonder if that was the issue, but I have
Same here.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Monday, April 25, 2005 10:14 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice Down?
I
Working fine for me..
going through: proxy.mia.broadvoice.com
if that helps..
--
Regards,
Sean Milheim
iDREUS Corporation
http://www.idreus.com
On Mon, 2005-04-25 at 10:27 -0700, Kerry Garrison wrote:
I seem to be down right now too.
-Kerry
-Original Message-
From: [EMAIL
I guess I didn't word this right.
It's not that SpanDSP ties up extensions, as it definitely doesn't. I
was more referring to the standard hardware-based solutions out there
that need to have a dedicated line for an incoming fax. I need the
ability to send and receive faxes with a good amount of
Max Clark wrote:
Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me it
cannot complete the call.
I can make outgoing calls from my system through broadvoice however.
Seems their inbound trunks hit capacity?
Am
Realtime voicemail configuration assumes the Voicemail Context to be 'default'
unless otherwise specified. This is not the same as the Extensions Context.
Having said that, can you specify what the actual problem is? Can't get
voicemail to pick up; MWI doesn't work; etc.
Matthew Boehm ([EMAIL
If you look at your iax.conf lines as under, you will notice that the
two contexts are illegal as they both have same name:
[telx-nyc]
type=user
secret=telx-nyc
context=from-telx-nyc
disallow=all
allow=ulaw
; telx-nyc-asterisk - Outgoing
;
[telx-nyc]
type=peer
username=telx-NY17S ; our
using Allison for the English prompts and are looking for
recommendations for Spanish.
You could check here: http://declic.com/voices/
There are 10 Spanish-speakers listed
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I tried calling Broadvoice support.. on hold for 1/2 hour then it hung
up on me with a reorder (fast busy), so I tried again.
Just got through to a rep- they said it's a 'carrier issue' that their
'partner carrier' was having issues and that it would be up soon.
Makes me wonder if I should be
And here.
BUT!!
I've spotted something odd. If I change the sip.conf settings as follows
from
host=sip.broadvoice.com
to
host=proxy.dca.broadvoice.com
I can receive incoming but not send outgoing.
Methinks they've changed something.
Mark
List Receiver wrote:
Same here.
I've had an issue with my 286 ever since I got it. Basically, the web
interface doesn't load, and I can't make any calls - although I get
dialtone. Also, I can call it and it will ring. But I get no audio.
The main issue is that I can't get into the web interface anymore... I
did once, but not
I read wrong. Outbound works fine. I am having same issues incoming.
On Mon, 2005-04-25 at 14:00 -0400, Sean Milheim (iDREUS Corporation)
wrote:
Working fine for me..
going through: proxy.mia.broadvoice.com
if that helps..
--
Regards,
Sean Milheim
iDREUS Corporation
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