products? Visit our VoIP store at http://voipstore.atacomm.com/
Atacomm can also provide you with competitive rates from your
local
carriers.
Remember: E-mail is not a secure medium. Please do not send payment
information via e-mail.
On May 12, 2005, at 1:27 PM, Wiley Siler wrote:
ipVolution
For free? FWD
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juanjo
Portela
Sent: Thursday, May 12, 2005 5:06 PM
To: Lista Asterisk
Subject: [Asterisk-Users] 1-800 free calls
Dear Sirs,
I was using iaxtel to make calls to 1-800 phones for free,
I have one working fine. The config is identical to the ones for IP500.
I wouodl look at my * setup. Is it new?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Sent: Thursday, May 12, 2005 6:23 PM
To: Asterisk Users Mailing List -
Personally, I always liked
TuxPBX.SomeDOmain.com
As for Greek and Roman Gods of communication...
Mercury for the Romans... Hermes for the greeks.
He was Zeus' messenger
For large growth systems, country names are very popular.
Otherwise, simple names of whatever fictional group makes
Have you done a debug of that SIP extension to see what happens when
someone calls in?
Also, are you using the config files for the Polycoms? If so, did you
disable callwaiting in the configs?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Go read the wiki. Look for MeetMe.
Ztdummy will serve as a timing device for machines without Digium
hardware.
Cheers,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Wednesday, May 11, 2005 1:32 PM
To: Asterisk Users Mailing
Title: IAX.CC/SixTel
Anyone have an opinion about these guys and their recent performance?
I need some local DIDs and they provide for my area code.
Thanks,
Wiley
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Rabii,
You should not double post if others have already answered
the original.
The previous answer pointed out that your description below
does not make sense.
Why would ATA2 call ATA2? It can call
itself? Do you mean "ATA2 can call
ATA1"??
Check your configs for the ATAs in sip.conf
Adam,
You should really look at [EMAIL PROTECTED].
http://asteriskathome.sourceforge.net
It has AMP and a ton of other features that will be useful
for a new user.
Cheers,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
CollardSent: Tuesday, May 10, 2005
Whew... What a relief.
I know the list was worried about why we could not get a hold of Manoj
Shetty
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manoj
Shetty
Sent: Monday, May 09, 2005 12:24 PM
To: asterisk-users
Subject: [Asterisk-Users]
Title: AAH 0.9
Is it possible to use the outbound routing features of AAH0.9 but also allow a user to dial a prefix to force the use of a certain route?
Thanks,
Wiely
___
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Outward dialing is a no brainer. VoipJet is the best outbound call
provider I have come across. Period.
It always works for me and the call quality is always very very good.
So far that seems to be the norm for them.
I am still working on getting my inward DIDs solidified so no opinion
there...
As a general rule of thumb it would be good to make the distinction
between 'Credit Card' and 'Debit Card' too.
If possible, never ever use a debit card for online purchases.
It taps directly into your account and removes REAL money.
Credit cards are 'virtual' money in that they are credit and
Contact me offlist but the basic premise is
this
You need only create an extension in sip.conf and then
correctly configure your phone.
If you want AMP, you should install Asterisk at Home
0.9
Sending script via FTP is thebest way to
go.
Via the web interface (you can get your IP by
Those are all three great phones and the choice gets really
preferential...
I love my Polycoms and I recommend them all the time. I give props to
the Cisco stuff but like you, I can't stand paying extra even if it is
just a few bucks here and there.
Polycoms can have a curve for figuring out
You should not have problems getting images if you come here or search
the Wiki.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, May 05, 2005 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I am a small customer myself and I have had some great service from
them. Granted, I have a had a couple of techs who were not particularly
polite or customer oriented in the past, however, they have been very
good with helping me. You have to realize these guys are growing so
fast and taking on
This winds up being very user specific. I love Polycom, others love
SNOM others love Cisco... Etc...
The Polycom IP300 is nice but only one row LED. IP500 is a great phone
and has 3 line LED.
Personally, I love the whole upright look and the phones have good
speaker phone.
$0.02.
Cheers,
Command line as headless Linux.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
pinchienSent: Tuesday, May 03, 2005 12:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
GUI
What is Asterisk GUI architecture acturally? I could not get
it...
The accoustic guitar collection here is pretty nice...
http://www.freeplaymusic.com/search/category_search.php?t=vi=41
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, May 03, 2005 10:59 AM
To: Asterisk Users Mailing List -
Well, that can be done...
Really should not do that though...
Cheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, May 03, 2005 12:22 PM
To: 'Matt'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
I highly recommend this isntall
Asterisk at home
http://asteriskathome.sourceforge.net
You have control of passwords so you can restrict some of the access to
the GUI stuff.
It utilizes...
Asterisk Management Portal (aka AMP)
http://amp.coalescentsystems.ca/
W
-Original Message-
My understanding is that Teliax may offer some unlimited dialing.
So far, my experience has been best dialing with VoipJet.
Your mileage may vary. You are best off to test a few and see.
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Command line on the box and navigate to the directory for your VM.
An example of one of mine...
/var/spool/asterisk/voicemail/default/1003/INBOX/
Issue the rm *.* command
Bye bye files
Your location may vary slightly depending on what * you are using.
I am on AAH 0.9.
Cheers,
W
What version of Asterisk are you using?
How were the music files transferred to *?
Some FTP programs default to ASCII and if you don't tell them to use Binary,
the file will transfer over with errors.
Done that several times and my MP3 files soudned horrible if they played at all.
Do you have
Title: Polycom IP500 - Phone TIme
Does anyoe know where I can set the timezone in the configuration files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields...
tcpIpApp.sntp.address=
: [Asterisk-Users] Polycom IP500 - Phone TIme
Wiley Siler wrote:
Does anyoe know where I can set the timezone in the configuration
files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I
enter this into the gmt fields in ipmid.cfg nothing seems to happen.
Here are the fields
on their Polycom set... There you go!
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 28, 2005 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
: Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Eric Wieling aka ManxPower wrote:
Wiley Siler wrote:
Does anyoe know where I can set the timezone in the configuration
files?
I am in Phoenix, AZ which has a GMT offset of -7 hours but when I
enter this into the gmt fields in ipmid.cfg nothing
Good points. I stand corrected.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Wade
Sent: Tuesday, April 26, 2005 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Extensions / Contexts
Wiley
Yep. I have this working now.
Thank you!
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
HalesSent: Tuesday, April 26, 2005 4:31 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] Polycom IP4000 Conference Phone
The
.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Clark
Sent: Wednesday, April 27, 2005 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Phone Recommendation.
Wiley Siler wrote:
Yep
I was afraid you would say that.
Does anyone out there have the latest firmware for the
Soundpoint IP 4000?
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
HalesSent: Monday, April 25, 2005 7:45 PMTo: 'Asterisk
Users Mailing List - Non-Commercial
-Users] Phone Recommendation.
How? You mean if you use [EMAIL PROTECTED] right?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
|Siler
|Sent: Lunes, 25 de Abril de 2005 02:50 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
Title: Polycom SIP 1.5.0 Firmware
Does anyone have it and can they make it available? My new IP 4000 will not work without it.
It is not currently posted anywhere that I can find from the Wiki.
Thanks!
Wiley
___
Asterisk-Users mailing list
Title: Polycom Config - SIP 1.4.1
Does anyone have an example of a working config for SIP 1.4.1? I made the transition and the phones seem to hate the new config file from the example.
W
___
Asterisk-Users mailing list
/resource_center/1,1454,pw-6812-9192,00.html
Original Message
Subject: RE: [Asterisk-Users] Polycom IP4000 Conference Phone
From: Wiley Siler [EMAIL PROTECTED]
Date: Tue, April 26, 2005 8:39 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Look at this code
;; IVR RECORDER; ; Record voice
file to /tmp directory exten = 205,1,Wait(2) ; Call 205 to Record new
Sound Files exten = 205,2,Record(/tmp/asterisk-recording:gsm) exten
= 205,3,Wait(2) exten = 205,4,Playback(/tmp/asterisk-recording)
exten = 205,5,wait(2) exten =
The short answer is No.
The method you describe is intrinsicly illogical.
Assuming there is an IVR, how will I know which extension 2000 I am
calling if that were possible?
I might get company A instead of company B.
You can create two * servers with identical dial plans, link them over
IAX,
Call waiting can be disabled in Asterisk via *71 regardless of the phone
used.
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean A.
Newton
Sent: Monday, April 25, 2005 11:56 AM
To: asterisk-users@lists.digium.com
Subject:
Title: Polycom IP4000 Conference Phone
Can someone verify that this phone uses the same configs and sip.ld and other files as the IP 500 ?
I jus tgot one and I cannot get it provisioned yet.
Thanks,
Wiley
___
Asterisk-Users mailing list
Seems to be the norm in most cases
I dial out on one ITSP who seems to always have good dialing with no
issues...
I receive on another that gives me DIDs and 800s (still working out
kinks)...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Who says on their site they are not taking new customers...
Bummer too, because I wam looking for a new set of DIDs from a reliable
source.
I was hoping they would be the ones
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Whitten
Sent:
Lower dial cost, cheaper redundancy, cheaper provisiojing (no $600 T1
Required)
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard
Marcel
Sent: Monday, April 25, 2005 1:46 PM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial
Inward or outward?
Outward, I am having the best luck with my dialing via
VoipJet.
Inward, I have nto stabalized any one service yet so I am
still open to suggestion.
Thanks,
Wiley
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD
AustinSent: Monday, April 25, 2005
Hmmm... Think I would prefer something harder to get provisioned but
that doesn't drop calls.
Your users must be forgiving as hell... Mine would show up with
pitchforks and torches if calls dropped regularly.
They get twitchy if the calls just vary too much in quality... 8)
Cheers,
Wiley
Just for grins A few thoughts.
Run Cat5 exclusively and just pull pairs for phone. Cheaper and better
solution that CAT3 and CAT5 mixed together.
It allows you to change the end points at will. Who knows if you may
want to change over to RJ45 ports and go total IP at some point.
You would
] On Behalf Of Wiley
Siler
Sent: Thursday, April 21, 2005 12:32 PM
To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
Hmmm... Think I would prefer something harder to get provisioned but
that doesn't drop
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, April 21, 2005 12:32 PM
To: Trevor Harrison; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] BYOD provider other than broadvoice
Hmmm... Think I
From your descriptions of your needs, you would be better served with an
AAH installation. Easier to understand than hand coding your contexts.
That aside, here are few answers...
Look here for more...
www.voip-info.org
Routing to the VoIP is just a matter of dial plan matching (see dial
I run a system with 2 cards with no issues related that I can tell..
Per the whole interrupt conver, 5 sounds like way too many.
Wouldn't something like this work though?
http://www.voicetronix.com.au/vpb4_v4pci.htm
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Interesting setup. Love to hear more about it as you get it done.
Regarding your DSP/PCI Bus I think that I saw that Sangoma cards may not
have the same problem.
Something to check out at least.
Also this card looks impressive if it ever materializes. Think it is still
vapor though.
Are you behind a firewall? If so, did you NAT an IP
to your * machine with a port forward for yourIAX
port?
Have you done IAX2 debug? Help iax2 should get you
the correct syntax.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
MasonSent: Thursday, April 14, 2005
Where is this line in zapata.conf under the [channels] context?
channel=1
Also is that line in zaptel.conf correct? Here is mine Note the
lack of and underscore on fxsks...
fxsks=1
loadzone = us
defaultzone=us
Try these settings and the run ztcfg -vvv
Restart * and see what you get
Title: Voicemail not working...
Hello All,
My voicemail seems to have stopped working and I cannot figure out why.
After call times out, the user receives a message the no one is available to take the call.
The CLI shows this...
-- Got SIP response 603 Decline back from 192.168.1.248
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
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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
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
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
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
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
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
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
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.
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