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
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
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
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
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
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
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
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:
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
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
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
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
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
Thank you both!
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Monday, April 04, 2005 7:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AAH 0.6 - Change Network Gateway
you can also
Actually, once you know what you are doing, the IP500 and
other Polycoms are quite easy to configure.
You just setup an FTP server to serve your configs off
of.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Monday, April 04, 2005 9:29
PMTo:
That worked gangbusters. After altering the ifcfg-eth0 file, a quick
ipdown and ifup of eth0 got me right where I needed to be.
Thanks!
How about the hostname in this version of AAH? I found one reference in
/etc/sysconfig/network but if I change the host in here, things break at
bootup. Is
Actually if memory serves, Vonage unlocks these for $10. Isn't that
true?
Not sure how that affects you price summation but it could be convenient
for those wanting to use these boxes.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
If you are using something like AAH then it is defined in the GUI and is
most likely *98 like the previous.
If you are hand coding these then you will need something like this in
your extensions.conf included in the context you want to have access to
VM.
[voicemail-secure]
Check your extension definition... Do you have a 'r'
in there to tell it to send ring tone back to the client?
exten = 3,1,Dial(1234,20,trf)exten =
3,2,Hangup
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
HanselmanSent: Tuesday, April 05, 2005 9:03 AMTo:
???
The Wiki is quite extensive I think...
Search the wiki for: Dial Plan
That should get you what you are looking for...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Tuesday, April 05, 2005 9:27 AM
To: Asterisk Users Mailing List
VoIP is a great solution if you can meet the minumum requirements
Data line - Can you get a line that offers synchronous 1.5 Meg
Bandwidth?
If not, the lower number on your up/down BW will be the limiting factor
for your calls.
Figure 80K uncompressed and as low as 20K (maybe lower?)
I see. I thought you meant that the upgrade was a
release from AAH. So there is still no upgrade path for AAH 0.6 to 0.8 I
assume.
As I understand it 0.6 backups cannot be restored to 0.8
systems of AAH?
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mr.
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
Run this from the CLI...
iax2 show registry
Do you see an entry that matches your LiveVoIP server IP (east or west
coast) and is it registered?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrejus
Stavickis
Sent: Wednesday, April 06, 2005 1:23
I have had far better luck than that too. More like a hour for me but
that is not too bad.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Wednesday, April 06, 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial
that they
have a west coast server (and gave me its IP to check routes to it) but
it wasn't quite ready yet.
Since I signed up though they pretty much stopped replying to emails
altogether.
On Wed, 2005-04-06 at 13:35 -0700, Wiley Siler wrote:
Run this from the CLI...
iax2 show registry
Do
Go here and look at 'i'...
http://www.voip-info.org/wiki-Asterisk+standard+extensions
Thanks,
Wiely
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Hewlett
Sent: Thursday, April 07, 2005 7:35 AM
To: asterisk-users@lists.digium.com
Subject:
I
signed up stating it is there west coast server located in Seattle. A
traceroute to that is only 8 hops and 40ms away from me. However if I
simply switch IPs in my .conf file, outgoing calls fail.
On Thu, 2005-04-07 at 08:26 -0700, Wiley Siler wrote:
I use the West Coast server
Yep.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, April 07, 2005 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Liveviop problem
I use the West Coast server. It is
The itsp I spoke with about concurrency limitations said they limited
due to overuse by calling card app providers.
By regulating the number of concurrent calls, they can maintain load and
quality for all users on the server(s).
Not being able to know your maximum line potential would be pretty
This is being covered from several different angles right now.
Google this: site:lists.digium.com DTMF Inline
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Heath
Sent: Friday, April 08, 2005 10:15 AM
To: Asterisk Users Mailing
What do you consider cheap?
At $400 I think these are cheap and new...
http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l
=enoc=sc420s=bsd
Cheaper then that?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent:
Francesco,
Asterisk is fully capable of the same performance and features as PBX
based solutions that cost 1000s of dollars more.
You just have to be willing to learn it, support it, and build it
yourself.
Please read the documentation at http://www.digium.com and
http://www.asterisk.org for
Tigerdirect.com often has refurb in your price area.
Walmart stocks a Microtel PC with Semprons for around $200.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty
Shackleford
Sent: Monday, April 11, 2005 12:13 PM
To: [EMAIL PROTECTED]; 'Asterisk
How about $80 then?
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?Edp
No=1271175Sku=P459-2001%20D
Froogle up a HDD and some memory and you should be in business.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Godee
Depending on how many users you want to support and price, there are
lots of options.
Smallest form factor will be SOC (System on Chip)
These are little more costly and not going to carry a huge load.
Next would be Mini-ITX
A bit bigger and will carry more load.
VIA is the king in this arena
If you have two lines registered to one phone then you need to do the
following...
This assumes extensions 1001 and 1002 are your line appearances...
exten = 1001,1,Dial(1001,20,trf) ;we are dialing line 1
-- After 20 seconds it will timeout and go to the next line
exten =
I have two cards installed in my AAH box. Once you
install, be sure to edit the zapata.conf (/etc/asterisk/) and zaptel.conf
(/etc/)
In
zaptel.conf
fxsks=1-8
In
your zapta set the channels to 1-8.
redo
the ztcfg - (these are vees)
Should
be golden...
W
From: [EMAIL PROTECTED]
What kind of performance does this system configuration give you?
Would it load out 20 calls with transcoding?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Latham
Sent: Tuesday, April 12, 2005 9:23 AM
To: [EMAIL PROTECTED];
Title: [EMAIL PROTECTED] - Newer Mobo - Memory
Hello All,
Are there any known issues with installing AAH on newer hardware that uses DDR2 memory and the latest mobos?
There are some SC420 servers out at Dell for $299 that are just beautiful for the cost and I would love to build my next
You just described a conference call which is supported by most phones.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of aram
Sent: Tuesday, April 12, 2005 6:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
If you have two devices on the same subnet and both are registered to *,
then calls will complete.
If the devices are on separate subnets, then you have to address issues
such as...
Firewalling?
Using NAT?
Routing in general?
SIP won't natively traverse firewalls so that would be a starting
As far as I can see, never gonna happen with an ATA.
ATA is your end point and has no exploitable features like that.
It just connects your analog phone to a digital network.
Meetme or Conference are probably your only bet in that case...
Check out AMP to see how call groups are used.
http://www.voip-info.org/wiki-Asterisk+Management+Portal
You group your phones, available handsets ring.
You can roll from group to group however you want.
Just a matter of writing the correct dialplan.
W
-Original Message-
From:
Title: IAXy Provision
Hello All,
Someone please tell me there is another way to provision an IAXy other than this horrid method.
http://www.digium.com/downloads/Iaxy_Installation_Guide.pdf
Thanks,
Wiley
___
Asterisk-Users mailing list
I have a neat PDF one that someone else authored that I can share.
Those interested, email me off list.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Wednesday, April 13, 2005 8:57 AM
To: Asterisk Users Mailing List -
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 3-Way Calling in Asterisk
Wiley Siler wrote:
As far as I can see, never gonna happen with an ATA.
ATA is your end point and has no exploitable features like that.
It just connects your analog phone to a digital network.
Meetme
And that is S much easier. Thank you!
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Wednesday, April 13, 2005 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXy Provision
Title: Polycom Vendor Recommendation
If anyone would like to contact me off list with the name of a good vendor of Polycom SIP phones, I would be most thankful.
I am looking to purchase 6 IP 500s and a Conference room phone. I am looking for a vendor who does good RMA and has excellent
Any chance that a version based Restore will happen anytime soon?
I would love to be able to restore onto a 0.9 with my 0.6...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, April 13, 2005 1:27 PM
To: Asterisk
/ertified on both IP and video products. will happily quote
great pricing.
- Original Message -
From: Wiley Siler mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
Sent: Wednesday, April
different
in0.9Maybe if I had a full time engineering staff! :)---
Wiley Siler [EMAIL PROTECTED] wrote: Any chance that a
version based Restore will happen anytime soon? I would
love to be able to restore onto a 0.9 with my 0.6...
W -Original Message- From:
[EMAIL PROTECTED] [mailto
Title: Wall Mount PC Case
Anyone have a recommendation on a good wall mount PC case that fits Matx?
Please respond off list.
Thanks,
Wiley
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
niggle entered my mind concerning ChanSpy - it was in use
at the time. I just can't think of how this could happen internally.
Julian
Wiley Siler wrote:
Do you share the same ISDN provider? Assuming all your VoIP is behind
a firewall and your only publicly exposed comms are across your ISDN,
how
Exactly. Time to check the CDR
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Thursday, April 14, 2005 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Overheard conversation.
Title: Bizarre - VM just stopped for one user
The other users work fine but this one does not.
Here is from the CLI on calling the user.
-- AGI Script Executing Application: (Dial) Options: (SIP/1000|120|tr)
-- Called 1000
-- SIP/1000-1bc2 is ringing
-- Got SIP response 603
Should have in iax.conf.
;This registers you to them
register=username:password@64.34.59.73
;THis context serves to ID incoming, if you ahve a DID
it shoudl come here
[livevoip]
type=user
secret=mySecret
host=64.34.59.73
callerid="Livevoip IAX
User"
context=livevoip-in
;This one is
:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent: Thursday, April 14, 2005 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Overheard conversation. Comments please !
Wiley Siler wrote:
The call bridge is the onoy thing that seems suspect.
Can
Just guessing but look for something like this.
This is from an old config of mine...
[trunkint] ; ; International long distance through
trunk ; exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9011.,2,Congestion
[trunkld] ; ; Long distance context accessed
through
Hello All,
I could use a recommendation if anyone has a
moment. I have the T100P but I have not gotten my service yet. I
want to have at least 12 lines of digital voice with DID. Should I just
seek out a PRI ISDN provider or is there something else I should look for?
I want to keep cost
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 1:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium T100P T1 Card
On Wed, 2005-01-05 at 01:01 -0700, Wiley Siler wrote:
Hello All,
I could
] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card
On Wed, 2005-01-05 at 05:59 -0700, Wiley Siler wrote:
Apologies if the format of the email was troublesome. I am
-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card
On Wed, 5 Jan 2005, Wiley Siler wrote:
So, I need to learn more about voice T1s? Reeally? That would be why
I am posting to the user group in the first place. To learn more.
The wiki says nothing about how PRI works
all,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Wednesday, January 05, 2005 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card
On Wed, 5 Jan 2005, Wiley
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Timothy
Costello
Sent: Wednesday, January 05, 2005 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium T100P T1 Card
On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote:
snip
To further explain my
Tim,
Just confirmed with ISP that the NIU connects to the AdTran over HDLC.
Thanks!
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, January 05, 2005 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, January 05, 2005 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Digium T100P T1 Card
On Wed, 2005-01-05 at 10:23 -0700, Wiley Siler wrote:
LOL
Jan 2005, Wiley Siler wrote:
A consultant so I can get a T1 PRI on my wall and use it with my
Asterisk box? LMAO. That is the dumbest thing I have ever heard. I
need a consultant so I can get a T1 with PRI? Please. I am just
trying to better understand how the Digium PRI card works
I have these very phones and took me a while to figure this out myself.
The phone considers each line registration to be a line with a second
line. So, call line while someone is on a call and another instance
will appear below. That means you only need one registered instance
for the phones to
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Adam,
Tor sent this one a little while ago that looks really promising for
solving the problem.
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tor Setane
Sent: Thursday, January 06, 2005 2:09 AM
To: Noah Miller
Cc: Asterisk Users Mailing List -
] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
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