Sys Admin wrote:
After 20 posts, in 2005 the ideal setup for a new installtion of a 50
user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
Seems like we cannot come to a definite conclusion,
Out of interest why use a G5 over an x86 PC? Do you feel the performance
will be better, or do you just prefer Mac's?
Thanks
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geoff Nordli
Sent: 21 March 2005 20:54
To: 'Asterisk Users Mailing List - Non-C
Asterisk wrote:
Try username=guest pass=emailaddr.
Nope:
220 Welcome to the Vink Consultancy FTP server. Please login...
Name (ftp.vinkconsult.com:brianc): guest
331 Password required for guest.
Password:
530 Login incorrect.
Login failed.
B.
___
Asterisk
Cool! I'm still away from the office, but I was starting to
work towards syching meetme2 up to the version of meetme in
* 1.0.7. It is over a 2000 line diff, ignoring the database
integration code, so it was looking like a not too trivial
task.
One question though, how difficult will it be to ex
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required - if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call with
low latency network (around 300 ms). I have included the debug
who has purchased a V400 card from Varion ?
if who has purchased V400 card
mail to me,
[EMAIL PROTECTED]
I am very appreciate.
I need some help .I need some document .
please help me .
thanks a lot
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Right,It looked there was a bug in the FTP server.I recompiled it, and assigned a password to the guest account.User=guest, password=restricted.This account wil be open util friday.Beware, the file is patched for the unstable branch.Andre- Oorspronkelijk Bericht -Onderwerp:ÂRe: [Asterisk-U
Hi all,
I'm new to asterisk and had just install it on my linux
server.
Can anybody told me how to setup it up for interworking with
cisco h323 voip gateway?
I check throught the manual on http://www.digium.com/downloads/marketing/asterisk.pdf
but cannot find any information for con
aram wrote:
Hello,
We are getting error: Call rejected: 407 Proxy Authentication Required - if
a user is trying to call using * over a long latency network (around 600
ms). There is no problem when the same user is trying to make a call with
low latency network (around 300 ms). I have included
For all who are interested: A quick review of CeBIT 2005. :-)
CeBIT was a very successfull event. Most of the time, the asterisk-booth was
crowded with more people than we could talk to.
We had with us a demo-installation including different IP-phones, digital and
analog phones as well as a Sieme
Hi Aram,
Think you will find it is something to do with ports etc
We log soft phones, spa-841, grandstream and sipura ata through a
satellite link that regularly drifts out to a latency of 1000ms ++ and
do not see any problems with *.
However some of the links we have used are into port bloc
Hello,
We are getting error: Call rejected: 407 Proxy
Authentication Required – if a user is trying to call using * over a long
latency network (around 600 ms). There is no problem when the same user
is trying to make a call with low latency network (around 300 ms). I have
included th
Hi Thorben,
Did you manage to take a look at I problem I described earlier ?
If a phone has more than one active call, only one call (the last one
received)
can be transferred. In attempt to transfer a remaining call IPSwitchBoard
will actually make a new call the number you were attempting to tra
Geoff Nordli wrote:
Hi Everyone.
Asterisk is one of those applications that need to be built from cvs on a
regular basis to keep up with the changes. I have always used package
management tools like apt.
How does everyone manage their Asterisk servers?
Geoff
Geoff,
Probably the easiest way is to
I also have a patched version on www.fitawi.com/Asterisk
The patch process started to grumble when I split-up the
patch file,
so I just stuck the patched version on the server.
The patch files are
available if anyone is interested in how trivial the patch
actually is.
These versions have
hi!
u can find more info here www.voip-info.org
/madhawa
On Tue, 22 Mar 2005 16:51:20 +0800, raymond <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>
> I'm new to asterisk and had just install it on my linux server.
>
> Can anybody told me how to setup it up for interworking with cisco h323 v
Hello,
I have come up with a php script that I believe should be able to send sms
alerts to cell phones as specified. I have added an option under the
extension account settings (am using [EMAIL PROTECTED] 0.6) successfully.
I would like to embed this script into the various specific configuratio
Hello,
I fixed this problem for me with some asterisk patching.
You can download patches at b2bua.berlios.de.
Short explanation: new option 'O' in Dial application will send only 1
codec (same as incoming) in outgoing invite. Curently only SIP channel
patched.
P.S. I'm not really good in asterisk
On Tue, Mar 22, 2005 at 02:07:32PM +1200, Cameron Beattie wrote:
> I have created a call file which has been moved into the outgoing
> directory. However the log file displays the following message: Unable to
> open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting
>
> I have exec
check b2bua.berlios.de
It AGI script for asterisk that support radius and many other fancy things.
On Fri, 18 Mar 2005 01:18:27 -0800 (PST), Kamran Ahmad <[EMAIL PROTECTED]>
wrote:
> hello pongco
>
> if you are talking about disconnecting a call session
> at his credit time. then you have to lo
Word of your booth came back faster than your mail ;)
Only good things where said ;)
/Danny
On Tuesday 22 March 2005 04.06, Thilo Rößler wrote:
> For all who are interested: A quick review of CeBIT 2005. :-)
> CeBIT was a very successfull event. Most of the time, the asterisk-booth
> was crowded w
ï
User=guest, password=restricted.This
account wil be open util friday.Nope:220 Welcome to the Vink
Consultancy FTP server. Please login...Name (ftp.vinkconsult.com:brianc):
guest331 Password required for guest.Password:530 Login
incorrect.Login failed.Yep:
$ ftp ftp://guest:[EMAIL PROTECTED]
I got the same trouble on some Zyxel ATA box ( NAT and non-NAT)
since I upgrade from 1.0.0 to 1.0.7.
It was working well with 1.0.0 but not with 1.0.7.
I am around 600 ms away from the * server.
Any Clue ?
but it has nothing to do with port blocking , Stateful , and so on .
the access I use , w
I've had 50+ people download the web components, and other
than reports of compile issues, I have not heard if this
collection has worked for anyone.
I do plan to keep updating the * applications and the web
pages, but I have almost meet all of our internal requirements
and wonder if anyone else i
You should have found my post with the exact same problem over a year
ago...
Oops. I think I found it but I thought "this patch is some months old; the
problem must have been solved in upstream versions long ago". My fault.
Apply this patch:
diff -ur zaptel/zaptel.h zaptel.mine/zaptel.h
I had to l
Hi!
> seems to like the Hold Pickup model. If you don't know what I mean by
> Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone
> and dial 12345 to pick up a call holding on ext. 12345.
>
> It looks like the closest to what I want (without changing Asterisk)
> would be P
> Word of your booth came back faster than your mail ;)
> Only good things where said ;)
Nice to read this :-)
Who did tell you about it?
--
Thilo Rößler
Linup Front
Robert-Koch-Strasse 9
64331 Weiterstadt
Tel: 06151/9067-0
Fax: 06151/9067-299
Mobil: 0151/18242584
http://www.linupfront.de
E-
Greetings *`s,
I am manually creating call files and dropping them into
/var/spool/asterisk/outgoing to be picked up by *.
Presently, when I use local/internal parameters using SIP it works..ie I
make an internal call from device to device.
However, when I try dial an outside number which I have
Greetings *`s,
I am manually creating call files and dropping them into
/var/spool/asterisk/outgoing to be picked up by *.
Presently, when I use local/internal parameters using SIP it works..ie I
make an internal call from device to device.
However, when I try dial an outside number which I have
Hi:
I was wondering whether there's a way to bridge two conference bridges
using Asterisk.
I want to allow a meetme conference to join an external conference
over the PSTN. One way of doing it, in theory, would be to use Omniis'
TAPI driver and place a CAPI call to the ISDN line (external conf.),
Hi,
Know there has been numerous posts on the subject of asterisk-addons and OS X. We have other uses for MySQL on the machine so changing over to Postgres at this point (which apparently works for CDRs) is not really an option. Have also contemplated a cron job to simply poll the csv cdrs and i
Thanks Mark will try that out!
-Original Message-
From: MF Hulber [mailto:[EMAIL PROTECTED]
Sent: 22 March 2005 05:25
To: Reuben Grech
Subject: [info] [Asterisk-Users] :: BIOS Motherboard Settings ::
I have the same motherboard. I put the card in the 2nd slot from the
bottom. In this
Hi
I must to estimate the* performance.
I am try to understand which can be the eventual bottlenecks.
Have you some suggestion?
Can you to signal to me some problems?
Thanks
Alessandra
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http
Hi Julien!
Julien Goodwin wrote:
On Mon, Mar 21, 2005 at 01:03:52PM -0700, Kevin P. Fleming arranged a set of
bits into the following:
Remco Barende wrote:
Are you sure? This is in the makefile:
# Asterisk version, currently only v1_0 and HEAD are supported
ASTERISK_VERSION=v1_0
Well, then the co
Hi Ivan,
I know it's a problem and I will look at it.
Thorben
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] På vegne af Ivan Meic (Vox Mundi)
> Sendt: 22. marts 2005 10:12
> Til: Asterisk Users Mailing List - Non-Commercial Discussion
> Em
On Tue, 2005-03-22 at 10:49 +0100, Fernando Sanchez wrote:
> > You should have found my post with the exact same problem over a year
> > ago...
>
> Oops. I think I found it but I thought "this patch is some months old; the
> problem must have been solved in upstream versions long ago". My fault.
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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google
On Mar 22, 2005, at 12:59, Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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I would say better hardware in the fashion of cooler hardware and
longer lasting. No flame wars or anything but, some CXXs have seen
Apple hardware in use for longer than five years on desktops (note
that they don't look arount in the NOC) and request it by name. Other
wise I would take an ISeries
before inventing the wheel once more, I would like to ask the list
Problem:
the dial command does it's job by
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],60,tTr)
now, if the given number (extension) is one of these
+49(2244)870663 or +49(02244)870663
the dial command fails, since it cannot handle "
Hello i need to make a central phone book, at this time we got a lot of
offices far away from here and i want to know if it possible to get a
central phone book from ldap or mysql to make calls just typeing the
name of the office, i saw the macros extensions using ldap but just to
get the caller id
CPU, RAM, Network
CPU - Translating signals of CODECS can use the CPU.
example: analog->TDM->GSM would take CPU usage and after many users (20+)
it would degrade on a 800mhz CPU
RAM - A minimum of ram should be used. I would suggest 256mb. After
the minimum everything else is just bonus.
Networ
On Mon, Mar 21, 2005 at 09:36:53PM -0800, Geoff Nordli wrote:
> Hi Everyone.
>
> Asterisk is one of those applications that need to be built from cvs on a
> regular basis to keep up with the changes. I have always used package
> management tools like apt.
I am known to always stick with the pack
On Tue, 22 Mar 2005 14:16:54 +0800, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> I have three different time displays:
>
> Flash panelcaller 615 48:00
> called 62058:18
> Snom phone shows for the same call 47:55
>
> Why is there a difference at all?
If you rel
> What would be a good combination to use on dialup connections? I know
> iax is better than SIP, but I dont' know much of anything about the
> various codecs. Also, how well would an iax or sip solution work
> compared to skype as far as voice quality?
I have a relative that is on dialup and cal
rfc3261
http://www.faqs.org/rfcs/rfc3261.html
Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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Hi all,
I want to know whether the following functions are already present in Asterisk (on SIP).
1.) To redirect a end-to-end call into a meetme room by pressing some key by either of the user (like Redirect Manager command)
2.) During a conference, invite more people by dialing their URI's wh
Excuse my ignorance here, but I am desperately trying to isolate the IRQ for
my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled all
usb , parallel, serial and some other devices, those that I needed to keep,
I have moved off of IRQ 10 and onto IRQ 5, but everytime I boot up,
Hi,
we've installed te110p with Suse 9.2 on Siemens primergy. We're connecting
to voxsteam i60 to test PRI interface.
We have problems, after reboot sometimes it goes green, otherwise stays
blinking red.
How could we debug this situation ?
Are there any common advices what to check ?
Are CVS He
Does it recognize the apic ?if not, do you run a multiprocessor kernel ?Try a multiprocessorkernel if you not run one already.it solved my apic problemsAndre- Oorspronkelijk Bericht -Onderwerp:Â[Asterisk-Users] IRQ headachesAfzender: ÂBrett, Gary <[EMAIL PROTECTED]>Aan:Â"'Asterisk Users Ma
Have you tried moving it to a different pci slot?
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A9.com: http://a9.com/SIP%20response%20numbers%20
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: Tuesday, March 22, 2005 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP resp
> Has anyone on-list had a chance to test these with * recently? I keep asking
> about once a month hoping that someone has moved forward in using this
> device. They look like great hardware, but early reports indicated that the
> firmware questionable. Still, that was 6 months ago.
I have two o
> -Original Message-
> From: Brett, Gary [mailto:[EMAIL PROTECTED]
> I have moved off of IRQ 10 and onto IRQ 5, but everytime I
> boot up, I get
> usb-uhci and ehci_hcd using IRQ 10 as well as my Digium card.
> Does anybody
> know what these are and how I can get rid of them ?
They're
Dear all,
1.Does Asterisk support SS7 and
ISDN?
2.Does Asterisk support SIP based
conferencing,audio ,video mixing
3.What SIP methods asterisk supports to enable SIP
based PBX kind of services??
Thanks in Advance
Wipro
Confidentiality Notice
The information contained in this el
On Tue, 22 Mar 2005 [EMAIL PROTECTED] wrote:
> 1.Does Asterisk support SS7 and ISDN?
ISDN is supported out of the box. SS7 support is (or will soon be?)
supported by a commercial version of Asterisk. Search the list archives or
post to asterisk-biz.
> 2.Does Asterisk support SIP based confer
I sent a few days ago right config parms that I got from BV. Try it, works
on my *.
Eugene B.
- Original Message -
From: "Jay Carter" <[EMAIL PROTECTED]>
To:
Sent: Saturday, March 12, 2005 5:42 PM
Subject: [Asterisk-Users] Broadvoice outgoing problems
Hello All,
I'm just ge
On Tue, 22 Mar 2005 12:10:17 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
> On a LAN where NAT is not an issue I would go for SIP + decent
> hardphones with good echo cancellation.
>
> On the internet with all sort of NATs + Firewalls, IAX is a must but
> unfortunately I don't know of any g
I'm trying to setup the TE110P on a multi processor machine but have a
problem too.
The console is reporting that the TE110P gets IRQ 0 and it suggest that
the MP table is faulty. Obviously the card doesn't work.
I will try the card in a single cpu box.
On Tue, 22 Mar 2005, Asterisk wrote:
Doe
http://asterisk.org/index.php?menu=features
On Tue, 22 Mar 2005 19:40:40 +0530, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Dear all,
>
>
> 1.Does Asterisk support SS7 and ISDN?
>
> 2.Does Asterisk support SIP based conferencing,audio ,video mixing
>
> 3.What SIP methods asteri
> Excuse my ignorance here, but I am desperately trying to isolate the IRQ for
> my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled all
> usb , parallel, serial and some other devices, those that I needed to keep,
> I have moved off of IRQ 10 and onto IRQ 5, but everytime I bo
Guido Hecken wrote:
...
this would change +49(2244)870663 to 002244870663 in every line of the
file,
named number.
But how can I achieve this in asterisk dialplan?
...
It sounds like an AGI perl script would work well.
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As
Adam Robins wrote:
I recently switched from BroadVoice to VoicePulse Connect on my Asterisk
box. Too many Meetme quality complaints (whether real or perceived).
I had to make a choice to use IAX2 or SIP with VoicePulse. I first
tried to go with SIP because I already had it working and all of our
Can Asterisks properly handle outbound Enhanced 911?
Blake
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Looking for a liitle help if anyone has dealt with this;
The options on dial and queue of t (allow called party to transfer call)
and T (allow calling aprty to transfer call) seem to work fine (as long
as you do not confuse them with the same t and T that indicate
timeout!).
The problem I am havi
I also noticed that. I'm running the 8/1/04 CVS version quite well over ny
global satellite network with 600-800+ ms latency (jitter of +/- 30ms). The
only time I've had an opportunity to try a new version (1.0.4) I found that
the satellite based SIP phones didn't work. I didn't have time to deb
I think your question needs to be "can my dial tone provider handle
e911?"
e911 is implemented by the party providing dial tone on the PSTN, at
some point your call must be routed to the PSTN to reach 911.
When you call 911 a dip is done into a psap database to retrieve the
address data and it is
On Tue, 2005-03-22 at 12:07 +0100, [EMAIL PROTECTED] wrote:
> Hi
> I must to estimate the* performance.
> I am try to understand which can be the eventual bottlenecks.
> Have you some suggestion?
> Can you to signal to me some problems?
Are you going to share your telecom engineering degree with u
Parker, Blake (MIS) wrote:
Can Asterisks properly handle outbound Enhanced 911?
Can the Ford F150 handle blue?
Neither of the above question makes any sense without additional
information.
Asterisk supports one of the 6 or so ways a PBX can support E911.
If you provide the details of what specifi
Previous answer did not address GPS coordinate issues, are your * users
mobile?
Part of the e911 service is to provide gps coords from the cell site (or
handset if so equipped). This information is useless for a stationary
user, the address is what is needed in this case.
> -Original Message
Jose R. Ortiz wrote:
Greg Boehnlein wrote:
On Fri, 18 Mar 2005, Jose R. Ortiz Ubarri wrote:
Jose R. Ortiz Ubarri wrote:
Hi:
I had asterisk with RealTime database working perfectly in a RH
9.0 machine. I used the sip cache so I even had MWI working. The
problem is that I decided to move
I'm in the US, using cards bought direct from Digium.
I have lowered the rxgain and txgain to -8 and that seems to be helping
futher.
I wish I could understand why?
The problem with more time is that I can hear myself in the headset of the
std. phone
as well as the party on the other end. The o
I want to be able to handle E911 from a service provider prospective.
Many customers from many different addresses and being able to properly
route the 911 call.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling aka ManxPower
Sent: Tuesday, Marc
My users will be stationary businesses
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Tuesday, March 22, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Enhanced 911
Previous answer
> > this would change +49(2244)870663 to 002244870663 in every line of the
> > file,
> > named number.
> >
> > But how can I achieve this in asterisk dialplan?
> > ...
>
> It sounds like an AGI perl script would work well.
Thanks for the tip with AGI I'll have a closer look at it.
Neveretheless I
Parker, Blake (MIS) wrote:
My users will be stationary businesses
There is not currently a good solution for doing what you want to do,
but there are possibilities in the works. The most likely candidate at
this time seems to be Intrado's V9-1-1 service.
__
Does anyone have any experience with asterisk and this radius module?
http://appradius.minitelecom.org/
If not, what radius module is recommened, for tracking SIP phone calls
for things like billing per phone?
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Just got off the phone with Net2Phone; they now require 3 credentials to
authenticate: account id, pin number, and MAC address. Any ideas?
Thanks
Russell Handorf wrote:
I can cut and paste the log file from a reload right now, and provide
you with the other information when I get home after work
Voip supply has a few 24 port gateways that are FXS based. The biggest
one for FXO is 10 ports. They are not cheap the both cost about $2000
USD. a Channel bank with a T1 card will cost you about the same at
least with a FXS ports.
FXO costs more usually because that is typically the Office sta
Before I go and try to write something myself, I'm curious if anyone has a
script that they're using for setting and clearing the MWI on a legacy PBX.
I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to
clear it, where XXX is the extension number. One complication is that
> After 20 posts, in 2005 the ideal setup for a new installtion
> of a 50 user asterisk is:
>
> Option1: IAX2 with softphone firefly
> Option2: SIP with softphone
> Option3: IAX2 with hardphones (which brand?)
> Option4: SIP with hardphones.
As the other poster said, I doubt you'll find a consensu
I have Festival running fine on one Fedora Core 3 machine but I am
having problems getting it to work on another one.
I am using festival-1.4.2-25
I have followed the guide at
http://www.voip-info.org/wiki-Asterisk+Festival+installation and am
using the second festival command patch which is the sa
I forgot to add that the problem I am experiencing is that when I dial
the extension it is answered and then immediatly hung up on me. It is as
if festival is working butnot generating any sounds.
On Tue, 2005-03-22 at 15:50, Gareth Blades wrote:
> I have Festival running fine on one Fedora Core 3
I have a SIP account that I can successfully register with
XTEN and a Sipura-2000. I have yet to be able to get it to authorize with
*.
My XTEN looks like:
Username: 001234
Password:
Authorization Username:
The feedback we are getting so far has been excellent! As more is
decided the list will be updated, if you'd like to be involved in
helping, please join us on the IRC channel, #asterisk-uk on
irc.freenode.net.
If your company would like more involvement with the event, please email
me directly. I
Actually, I love my install of AAH 0.6.
When something is not available in AMP I just dive into the configs and
correct it.
Most of the little things ARE available in AMP though so those times are
few...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behal
Absolutely... Mine was just a passing joke and not intended to make
Raphael feel bad.
I really did think it was quite funny. Such is the way of cross
language requests.
Sometimes the meanings get muddled and are humorous.
Hopefully the documentation gets him start on the * path and we hear
from h
Situation:
New Install of Asterisk
7960 w/ SIP 7.4 Image
7912 w/ SIP040406A
3 Lines Defined on the 7960
(5104,3100,2100)
Questions (configs are below):
Why won’t the MWI light
on the Cisco? I’ve tried:
mailbox=2100
[EMAIL PROTECTED]
[EMAIL PROTECTED]
D
For OH323 there is a workaround
Before dialing out, do in your dialplan :
exten => XXX,1,SetGloabalVar(OH323_OUTCODEC=g729)
We are also preparing a version that has endpoint configuration
like in sip.conf. It will be ready soon.
George
Mike Tkachuk wrote:
Hello,
I fixed this problem for me with som
There was a bounty a while back to set up SMDI on *.This would be ideal
if you had a serial interface on your PBX. By the #63 and #64 code it looks
like you are talking about a Toshiba PBX. At one time I actually wrote a
cron script that would check to see if there were messages in folders
Hi,
With everyone other that who uses Asterisk.. what is the best solution
you have found for billing VoIP users? Radius? Just parsing CDR
reports nightly?
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This is a common issue with all Sipura devices I've seen. I set the
registration interval to 5 minutes, so that NAT doesn't interfere.
I've done this with Cisco IP Phones, Cisco ATA Converters, and Sipura
SPA-1001, SPA-2000, SPA-2100, and SPA-841's. All Sipura's have the
same issue, 1 out of every
I've been having the same problem... The trouble is, the version of
gcc that Apple releases for OS X does not support the "-shared"
option, so any makefile directive which uses that can't be built.
This includes zaptel, libpri, format_mp3, and maybe more. I don't
know if there is a workaround...
I've been trying to get a new asterisk box setup with Broadvoice for
over a week now.
I have it connecting and registering with them according to 'sip show
registry',
I can't dial out through it, but it does dial out through my regular
phone line.
I'd like to set it only to dial 911 through tha
Do you have the broadvoice trunk set as the Default Trunk?
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Tuesday, March 22, 2005 8:33 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Setup to dial out only on voip
Kerry Garrison wrote:
Do you have the broadvoice trunk set as the Default Trunk?
-Kerry
Looks like I have more reading to do :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD Austin
Sent: Tuesday, March 22, 2005 8:33 AM
To: asterisk-users@lists.digi
Thanks, I don't play with web pages to much. It has a lot of great stuff
for a newbe like me.
Thanks, David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Tuesday, March 22, 2005 8:01 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
the topic says it all really.
Does the Sipura 3000 detect and report UK clid correctly?
thanks
Mike
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> I've been having the same problem... The trouble is, the version of
> gcc that Apple releases for OS X does not support the "-shared"
> option, so any makefile directive which uses that can't be built.
I have never done any OS X compiles but
http://fink.sourceforge.net/doc/porting/shared.php?ph
Although some people may disagree with me on this list. I think for people
new to Asterisk, it is often best to start with the [EMAIL PROTECTED] build to
learn the ins and outs a little easier and then moving away from the @Home
interfaces when you need to add additional functionality that you cant
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