Why would one want to do this?
If segmenting the application for several businesses is your goal (I am
guessing) then it just comes down to dial plan and context management.
There would be a huge amount of resource contention that would occur
with the system you describe.
W
-Original
I just dropped my X-lite phone for the SJPhone.
It was a hrd choice and one that was suggested to me repeatedly before I
would give in.
My reason being that the X-Lite phone is very user friendly and worked
well sometimes.
After switching to the SJPhone I am much happier with the call quality.
provider does not make the
product available in an unprovisioned state. Are you sure this applies
to the Linksys or are you referring to a situation where Vonage unlocked
a Cisco or Motorola ATA?
On Apr 5, 2005 8:14 AM, Wiley Siler [EMAIL PROTECTED] wrote:
Actually if memory serves, Vonage
Go look on the wiki for DISA. That should be averything you want.
Direct Inward System Access
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob
Cazzell
Sent: Monday, April 04, 2005 1:05 PM
To: asterisk-users@lists.digium.com
Title: AAH 0.6 - Change Network Gateway
Hello All,
For this CentOS based release of Linux and Asterisk, where are the networkign setting saved?
I need to change my gateway but so far I have been unsuccessful. Is there a tool for this?
Thanks,
Wiley
How can TOS tagging on the IAX channel affect a phone that is completely
SIP?
In my experience, the robot voice issue usually arises when bandwidth
restriction or latency occur on the data line providing the IAX call.
Assuming your connection is like this
Sixtel IAX --- Asterisk Box ---
Many options
http://www.esis.com.au/SmallPCs/Compact_PC.htm
System-on-Chip
That is a term you want to look at for really small items...
IT does not hit your 500MHz but it is the smallest thing I have ever
seen...
http://www.norhtec.com/products/gp/index.html
Other than that, you can just
See my other email
My setup
VoIP Provider --- My T1 --- Asterisk --- Sip Clients
The only time I get robo-voice is when the latency to the VoIP provider
is high.
Translating from IAX to SIP should not be a problem but maybe it is in
the build you have?
I run COS on my Polycom segment
Well, in the Asterisk arena, it allows a few nice things...
If you are on one call, you can see your second call come in on a new
line and choose to put the current on hold and answer it or just ignore
it. It allows you the ability to leave someone on hold at your station
while calling someone
Un what is todays date?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz
Sent: Friday, April 01, 2005 1:31 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
This version support SS7 -
Of Wiley
Siler
Sent: Friday, April 01, 2005 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What's the use of a multi line phone?
Well, in the Asterisk arena, it allows a few nice things...
If you are on one call, you can see your second call
Has the users hardware been assessed yet? I cannot remember seing
anything regarding the hardware for this issue.
I am sure memory and processor speed will play a part if lots of calls
are active during the transcode...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Are you sure you don't mean Ringtone?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Thursday, March 31, 2005 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Livevoip still no DTMF?
I
It is...
Polycom
456
The setup for using new confs and app files is done through the phone
anyway. Just setup the FTP server and your files.
Then at least you should be able to get the latest app file son the
phone to ensure it works right, even if not configured correctly.
W
Google this...: site:lists.digium.com
mini-itx
Lots of good info in the archive on this
one...
Wiki
http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware+mini-itx
For some nice mini-itx hardware examples check
out
www.mini-itx.com
Also, you may consider Micro-ATX if you want
Any possibility to support a zero extension and operator extension
automatically in the Auto-attendant?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, March 29, 2005 9:33 AM
To: Asterisk Users Mailing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Tuesday, March 29, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released
On Tue, 29 Mar 2005 09:43:04 -0700, Wiley
Read teh seciton covering AMP
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey
SharpeSent: Tuesday, March 29, 2005 1:43 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users] With a phone system.
I was looking thru the archives
Start here...
http://asteriskathome.sourceforge.net
Find a lnk to AMP
http://amp.coalescentsystems.ca/
All you need to get going is to got to the http://machine IP then go to
AMP.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey
SharpeSent: Tuesday, March 29,
Mar 2005 09:43:04 -0700, Wiley Siler
[EMAIL PROTECTED]
wrote:
Any possibility to support a zero extension and operator extension
automatically in the Auto-attendant?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Did you also include an entry in voicemail.conf?
After that the most common mistake is referencing a bad context for your
VM.
As long as you have it right, it should work fine.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robson
Ribeiro
Sent:
So far, in my experience, LD as we know it as POTS users is not the same
as LD via VoIP.
Lines over VoIP are marketed as anywhere to anywhere for minute costs
ranging from 1.1 to 2 cents or higher.
Whether you call next door or all the way to the East coast from the
West coast, the cost is the
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
, I guess. So be nice, gUys!
OK? Nevertheless, it was a good laugh :-).
--Luki
Wiley Siler [EMAIL PROTECTED]:
What a great way to end the day! This one has me laughing my ass
off
I am hoping you actually meant guys. You may want to look up the
meaning of the word you used
a pain cause I
have to put the port on my dial and registry entries in order to
register on it or dial to it.
Why is that?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Jueves, 17 de Marzo de 2005 03:08 p.m.
To: Asterisk Users Mailing List
What version of Asterisk? If this is not [EMAIL PROTECTED] you may want to
install it and start over. It eases many of the problems experienced by newbs
when learning *.
Otherwise, make sure you use the ztcfg - so you can see some error
verbosity.
You may need to recompile your zaptel
Contact me offlist and I will gice you some info
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bram
Sent: Friday, March 18, 2005 10:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie question
I installed an [EMAIL
If this is an AAH install, go to the maint section of AMP
and look at your configs.
Find where you *98 is defined.
This is in app-messagecenter context of extensions.conf on
my AAH 0.6 build.
We just need to reference this context as an include in the
incoming context for yoru dialplan
I think Cisco VoIP phones are absolute works of art. The first time I
saw one, I wanted them,
That being said, I use Polycom IP 500s and I absolutely love them.
The speakerphone is excellent, configs are pretty simple once you know
what you are doing with them, and the phone is very
I recorded my last set of prompts over my Plantronics DSP 500 USB
Headset. I have also used a Logitech USB Headset. These and similar are
easiest to use along with X-lite or similar softphone. I used the
suggested method of dialing an extension on the PBX and letting Asterisk
record for me direct
Fun things with MOH to remember...
Are the MP3 files you are using constant bitrate?
If transferred via FTP to the * machine, did you set Binary before the
transfer?
If this... (or similar)
exten = 6000,1,Answer
exten = 6000,2,MusicOnHold()
Gets you music you can hear, then the issue is
).
Tony Davidson CNA CA (IT) DCE
Director, Zero Effort Networking Pty Ltd
Ph: 0411 478 004, Fax: (02) 8569 2012
http://www.zeroeffortnetworking.com.au
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, 17 March
I cannot answer q1 and am interested in this myself.
Question 2 has a partial answer in that the AAH has a backup feature
located in the management portion of AMP.
The backup link is at the bottom. The restore feature is located at the
linux command line on a AAH machine.
help-aah will show
Gianluca,
Did you install the .59r. Version of mpg123? The most common problem I
have seen for this is that people keep installing the 59q or 59g version
of mpg123. 59r is the way to go.
http://www.voip-info.org/wiki-mpg123
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
Please check the Wiki (www.voip-info.org) and the list archive by
Googling site:lists.digium.com search string
Also, please include some more info. That is probably why you got no
answer...
Is your machine sitting behind a router or is it directly connected to
your broadband (assuming)?
If the
to an old address.
I'm not sure what you mean here. Which settings?
Thanks,
tony
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, 17 March 2005 6:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
What doesn't make sense about that is that if you are setup like this...
DSL Router --- Your Firewall/Router --- Asterisk Box
Then the issue of being dynamic will not matter to the * box. IP
storing is mute since the end point and start point are not changing.
All that is changing is the IP on
Type 'mpg123' at the Linux CL. (no quotes)
If the version is anything other than 59r, you problem is solved.
Go to the Wiki and search for Music On Hold.
Do the install of version 59r ONLY as described in the docs.
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
More info.
What IAX service are you trying to connect to from your * box?
So... Are you saying you have a DSL modem in a Smoothwall firewall that
routes between the DSL modem (eth0) and the NIC (eth1) that serves up
connectivity to your internal network?
Your configuration is getting more
, March 16, 2005 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] (Yet another) Music on hold problemand
another...
Wiley Siler wrote:
Type 'mpg123' at the Linux CL. (no quotes) If the version is anything
other than 59r, you problem is solved.
Go
This is what you need.
Google allows you to enter a parameter called 'site:'
when you do this it searchs that site only.
The list is archived so you always have it available.
Search at google with the following...
site:lists.digium.com some
parameter
This will search the archive and you
Better duck...
These kinds of request usually incur the wrath of the list since your
question implies you have not really researched this yourself.
Check Google first please
Google for this exactly...
Site:lists.digium.com insecure=very
The site: portion tells it to look at the list
Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
Excellent thing to
hear. I am glad there are positives on this site as well as teh
warnings.
Now to get the ringback issue
resolved
Using m switch to get MOH is OK but there
has to be alogical reason this is occuring adn a
Title: [Asterisk-Users] SIP to H.323 no audio
Yep. Of course, problem is the provider gave the
settings and the deny statement was part of it. Ooops to them i
guess.
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C.
TomlinsonSent: Friday, March 11, 2005
The port for sip is 5060.
Why no just map an ext to an internal and the problem us solved.
Assuming you have FW access and enough Ips.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Harold
Fletcher
Sent: Friday, March 11, 2005 10:55 AM
To:
And if getting service is your only concern then look at these
Teliax.com
Livevoip.com
Voipjet.com
There are more but those are I can think of right now.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Friday, March 11, 2005
Title: No ringback over IAX - LiveVoip
Hello All,
I saw some coverage of this in the list archive but no one seems to have posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR.
From there the call is routed to groups
http://www.voipsupply.com/product_info.php?products_id=456
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sylvain
COUTANT
Sent: Friday, March 11, 2005 11:42 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Wireless VoIP
Hi all,
I have
Which happens to not be covered as an unlimited service. You will have
to pay for minutes though some are included by default.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dalon
Westergreen
Sent: Friday, March 11, 2005 11:48 AM
To: Asterisk Users
I am a Polycom guy so IP300 or IP500 comes to mind.
Also this Grandstream looks interesting...
http://www.voipsupply.com/product_info.php?products_id=331osCsid=adc1f6
01a12939e2a97ca342d73173b7
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Title: Sip show registry returning nothing
Hello all,
For some reason I am not showing registration in SIP.
Can anyone give me an idea what can cause this?
asterisk1*CLI sip show registry
Host Username Refresh State
___
Asterisk-Users
Yep.
Search the Wiki for CDR and AMP
Wiki is here...
www.voip-info.org
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanishka
SomaratneSent: Friday, March 11, 2005 12:18 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
Billing
Why answer if you are bothered? Isn't that just burning more of those
electrons of yours? 8)
People get stuck on things and need a hand. Doesn't mean they are lazy
by default.
They just don't know better.
In this case, it is a good chance to direct him to other providers who
do what he wants.
a registered entry.
Bizarre
Thanks!
W
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Wiley Siler
Subject: Re: [Asterisk-Users] Sip show registry returning nothing
On Fri, 11
PROTECTED]
Sent: Friday, March 11, 2005 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Wiley Siler
Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip
On Fri, 11 Mar 2005 11:47:53 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
I saw some coverage
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Friday, March 11, 2005 12:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sip show registry returning nothing
Wiley Siler wrote:
Hello all,
For some reason
.
Then we can correct the person so they can be a GOOD submitter to the
list.
That way we all benefit.
Cheers,
Wiley
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 12:57 PM
To: Wiley Siler; Asterisk Users Mailing List - Non-Commercial Discussion
[mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 1:22 PM
To: Wiley Siler; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sip show registry returning nothing
On Fri, 11 Mar 2005 12:48:49 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
All my SIP phones
] Sip show registry returning nothing
On Fri, 11 Mar 2005 12:48:49 -0700, Wiley Siler
[EMAIL PROTECTED] wrote:
All my SIP phones are still working and all my dialing is still
working, so I did not think it relevent.
Sip reload...
asterisk1*CLI sip reload
Reloading SIP
== Parsing '/etc
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No ringback over IAX - LiveVoip
On Fri, 11 Mar 2005, Wiley Siler wrote:
I saw some coverage of this in the list archive but no one seems to
have posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call
PROTECTED]
Sent: Friday, March 11, 2005 1:37 PM
To: Wiley Siler; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Vonage a provider?
On Fri, 11 Mar 2005 13:16:23 -0700, Wiley Siler
[EMAIL PROTECTED] wrote:
How is this apparent, I would be glad to help you, when
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Goerzen
Sent: Friday, March 11, 2005 1:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Vonage a provider?
OK, I missed the message that started this thread, but:
On 2005-03-11, Wiley Siler [EMAIL PROTECTED] wrote
:
On Fri, 11 Mar 2005, Wiley Siler wrote:
I saw some coverage of this in the list archive but no one seems to
have posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from LiveVoip
over IAX I dump it into my IVR.
From there the call is routed to groups based upon
blown their profit.
I myself have often walked away from expensive customers, and business
people much smarter than me do this on a daily basis.
-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 1:53 PM
To: Robert Webb; Asterisk Users Mailing
.
Everyone agrees that it's on LiveVOIP's end, but they're shrugging their
shoulders and pointing toward *. Search the list.
-Original Message-
From: Wiley Siler [mailto:[EMAIL PROTECTED]
Sent: Friday, March 11, 2005 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello Androtech,
The issue you are having is by design. >From a
firewall stand point, you would never want packets coming in from the external
unsecured to terminate at the internal nic IP. That is
counter-intuitive. You might FORWARD that traffic somewhere
internalbut you would not move
I am buying and 800 number and dial time from LiveVoip today.
Cost is 1.2 cents per minute. I will let you know about quality when I
know more.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Thursday, March 10, 2005 4:06
You should go here and enter your suggestion so the developers can track
it.
http://sourceforge.net/tracker/?group_id=123387
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Glenn
Sent: Thursday, March 10, 2005 4:38 PM
To: [EMAIL
Title: [EMAIL PROTECTED] - Email to Fax
Hello All,
I am preparing to get Email to Fax setup on my AAH 0.6 box. While I am finding plenty of doumentation on how to get this done using AstFax, it occurred to me that I may want to ask if anyone knows the mechanism of how AAH currently
10, 2005 4:04 PM -0700 Wiley Siler
[EMAIL PROTECTED] wrote:
I am buying and 800 number and dial time from LiveVoip today.
Cost is 1.2 cents per minute. I will let you know about quality when
I know more.
Thanks,
Wiley
___
Asterisk-Users mailing
You need to give lots more. If othing has changed in the settings on
the Asterisk box, then look to your x-lite config, the firewall
settings, or connectivity (think port closures by an ISP or admin).
Best I can offer with limited info.
Cheers,
Wiley
-Original Message-
From: [EMAIL
Title: [Asterisk-Users] SIP to H.323 no audio
Hello all,
I am having trouble getting my IAX based Voip
provider setup. Any pointers are welcome.
So here is the deal. I am registered up and I can make outgoing calls but incoming calls
fail.Configs all look good I thought.My PBX is behind our
of ordering via the online
website. And that was at midnight on a Saturday night. Of course, they
don't guarantee that, I think I just got lucky... :)
Paul
- Original Message -
From: Wiley Siler [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Title: [Asterisk-Users] SIP to H.323 no audio
OK. I removed the deny
statement they have me using and now I can get incoming calls.
Do I need the deny 0.0.0.0/0.0.0.0 statement?
Thanks,
Wiley
From: [EMAIL PROTECTED] on
behalf of Wiley SilerSent: Thu 3/10/2005 11:59 PMTo:
Asterisk
Title: VoIPJet
Anyone suffering an outage with them right now?
I am getting the following from my box when I try to dial using them.
== No one is available to answer at this time
W
___
Asterisk-Users mailing list
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIPJet
On Wed, 9 Mar 2005 14:13:21 -0700, Wiley Siler
[EMAIL PROTECTED] wrote:
Anyone suffering an outage with them right now?
I am getting the following from my box when I try to dial using
them
You are correct. Apologies.
Wiley
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 09, 2005 2:47 PM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIPJet
On Wed, 9 Mar 2005 14:40:17
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: Wednesday, March 09, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIPJet
On Wed, 9 Mar 2005 14:13:21 -0700, Wiley Siler
[EMAIL
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: Wednesday, March 09, 2005 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIPJet
On Wed, 9 Mar 2005 14:48:50 -0700, Wiley Siler
[EMAIL PROTECTED] wrote:
You are correct
Someone else mentioned this card earlier and it looks
promising for a way to get your 4 E1s up and have DSP on the
card.
If I were provisioning today I would consider this one for
sure though I am betting there is no real data available
yet.
ipVolution TDM120
Better yet, ditch the Mandrake box and try [EMAIL PROTECTED] for you test
machine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, March 08, 2005 10:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users]
???
The only way a SIP client can connect to Asterisk is if there is an
entry defined in sip.conf. That unto itself requires passing the
extension name and the secret which is essentually username/password as
you are requesting.
Google sip.conf at the Wiki.
W
-Original Message-
From:
Port count should be the only net difference.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Spencer
Nassar
Sent: Tuesday, March 08, 2005 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Wildcard X100P or TDM400P?
I'm looking
Steven is totally right BTW. No support for clones.
However, I will tell you this. I built my first box with a clone so I
could see if I could do all I needed on this system. Then I promptly
purchased my two TDM400s for my 8 POTS lines. Wish it were PRI but
those are the breaks for this
-Original Message-
From: Race Vanderdecken [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 08, 2005 11:35 AM
To: Wiley Siler
Subject: RE: [Asterisk-Users] force SIP authentication
Florian,
The code basically allows any device that shows up to try and do a SIP
REGISTER.
Using a username
Title: STOP NOW not responding
Has anyone had any new information about STOP NOW hanging? I am using [EMAIL PROTECTED] 0.6 and today my system just stopped responding. I issued the usual STOP NOW command and it just returns to the CLI. I have found a lot of info regarding others having this
Welcome to the wiki located here...
http://www.voip-info.org/wiki-Asterisk
Also, refine your google search to include this at the beginning...
Site:lists.digium.com
That tells Google, to search only the pages from this email list.
Regards,
Wiley
-Original Message-
From: [EMAIL
:[EMAIL PROTECTED]
Sent: Monday, March 07, 2005 12:08 PM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Setting up asterisk with current PBX?
Thanks for the exceptionaly fast response. I got all the info I need
now!
On Mon, 7 Mar 2005 11
Check the can reinvite setting for NAT issues.
Check the wiki for how to configure as you have described.
http://www.voip-info.org/tiki-index.php?page=Asterisk
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
Ignore the error if it isn't messing anything up.
Check out the Wiki here
http://www.voip-info.org/tiki-index.php?page=Asterisk
A search of X-lite here also yields proper setup info for the softphone
to Asterisk connection.
The archive of this list can be search via google by entering...
configuration is pretty straightforward, you just give it
username/password and point it at a SIP proxy. However, as far as I can
tell it isn't able to register, or it's not listening to Asterisk...
hard to tell really.
-Brian
On Mon, 7 Mar 2005 14:32:57 -0700, Wiley Siler
[EMAIL PROTECTED] wrote
Do you have the handset still off the hook when you do
this?
If the handset is on hook and hit the speaker button it
should hagup the call.
If the handset is off hook, it should revert back to the
handset.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
Go here...
Type budgetone in the search box on left.
http://www.voip-info.org/tiki-index.php
W
From: [EMAIL PROTECTED] on behalf of Bill Michaelson
Sent: Sat 3/5/2005 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
, then I only
want to learn to configure the different config files, that's all.
Thank you, Do not be angry at me!
Willy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Friday, March 04, 2005 12:35 AM
To: Asterisk Users Mailing List
Also lookup AGI
The WiKi and via google by using this: site:lists.digium.com some
words
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, March 04, 2005 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial
What is your price range is the question. BudgetTones are OK but have
some limitations. Polycoms are my choice for around $160-$200. Ciscos
work well for some people too. Just a matter of how many dollars. The
budgettone lack of 3-way can be gotten around with a proper dial plan.
W
And here...
www.digium.com (see documentation link)
And when you have some early questions look here...
www.google.com enter this: site:lists.digium.com whatever you are
searching for
And here
www.asterisk.org
Read as much as you can...
W
-Original Message-
From: [EMAIL
Go to www.voip-info.org and search on the word CDR
http://www.voip-info.org/wiki-Asterisk+CDR+Areski+GUI
You can also search on the word queue
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Scully
Sent: Friday, March 04, 2005 3:37 PM
To:
Did you install the version of mpg123 that is 59r and not
59q or 59g?
This
is a problem with version of mpg1223 almost assuredly.
Did you install just as the installation says to do?
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
More stuff..
And it would not hurt to just ftp off your confs to another box for
safety.
Or use WinSCP3 if you want that GUI app feel to connect to * from
Windows.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Thursday, March 03, 2005 1:48 PM
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