One might also conclude that during the outage the support people were
focusing on getting the system back up and were not near phones. At
least that is what I would bet on. Just a thought considering how most
of the smaller ITSPs seem to work.
Cheers,
Wiley
-Original Message-
From:
your objective.
From: Wiley Siler [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
Date: Mon, 27 Jun 2005 08:30:41 -0700
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
LiveVoip has been
So far my experience with TOS has been that most of them are pretty odd.
No one wants the liability of a stock trade gone foul or a call to the
doctor that gets disconnected. Essentially, those things in the TOS are
just a CYA. They are un-enforced but should someone decide to attempt
to sue
Great points Steve. I think the best we can do is all throw the newbies
a bone ounce in a while. Redirection to the content that is relevant is
enough to get most people on the path. Like you said, the hardest part
is not seeing the trees for the forest.
This is the whole teach a man to fish
Subject: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]
On 2005-06-28, Wiley Siler [EMAIL PROTECTED] wrote:
So far my experience with TOS has been that most of them are pretty
odd.
Not THAT odd :-)
No one wants the liability of a stock trade gone foul or a call to the
doctor that gets
One would assume they have better things to do as they are quite busy.
I think this is just a proactive measure meaning they say you cannot do
it upfront so that in the event of a problem, it was predeclared. As to
the rest of the TOS, I could be wrong but I got the impression that the
owner of
You just got a tax write off because your money is certainly locked up
in chapter 11.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bashir
Ullah - www.Lamsre.Com
Sent: Sunday, June 26, 2005 4:02 AM
To: Asterisk Users Mailing List - Non-Commercial
As predicted
In keeping with LiveVoip company policy, even this letter seems
antagonistic towards customers and creditors. You are under a STAY!!
Don't talk to us!
Wow, I guess the merger with the trailer park DSL company just did not
help at all. And after Joop spent so much time
Well, I guess stating at the bottom of the list is a bad idea sometimes.
Sorry, Marcel, I just find this a relevant topic since so much money and
time have been wasted trying to use this company's service.
Will drop it shortly though.
Cheers,
W
-Original Message-
From: [EMAIL
Actually, if you look at my posts from a month or two ago, you can see
that they not only had to have known, they were publicly stating that
they were expanding. Joop personally told me that they were going to
offer Vonage type of service and that they were opening service in the
UK. He actually
LiveVoip has been a learning experience for anyone who purchased from
them. With any luck, it was a learning experience in what not to do
for anyone out there that provides similar services. At least I hope so
since it seems obvious that LiveVoip never learned a thing during their
interaction
Having read the TOS from LiveVoip many times, I can almost assure you it
was written by the LiveVoip staff and not a lawyer. Due to that, I
cannot imagine them slithering out of this entirely.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marie
This is probably a good time to point out that there is a good litmus
test for all Voip providers. PRIOR to purchasing anything, send them an
email and request the sales information. Ask about their servers or
their policies or anything you can think of. How they respond will tell
you a lot.
Well, as someone who doesn't use threads... I think I can say it is not
the end of the world. I find scanning my Asterisk mail folder to be
pretty easy
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, June 27, 2005 8:47
For outbound only, I have traditionally recommended VoipJet. They just
recently has a spat of issues that seem to have resolved though. I am
still using them via their east coast server and it seems to work quite
well again. Cost is around 1.3 cents minute I believe. Use IAX and
g711 for best
How about LinPhone?
http://www.linphone.org/?lang=usrubrique=1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hamish
Whittal
Sent: Monday, June 27, 2005 12:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] OT:
Feel free to unsubscribe at your earliest convenience.
I for one disagree with you completely.
The only questions that occasionally go unanswered are those that were
just asked, are written twenty times on the Wiki, or are intrinsicly
stupid.
And even those questions are often answered with
The archive is very searchable it just is a pain sometimes...
To search by relevance...
Go to google.com.
Enter: site:lists.digium.com some search value
You get the archive back with relevance as the main sorter.
I have also seen some places that back up the list and sort by date and
allow
Well, we could aspire to be a great list in his eyes you know. (LOL)
Maybe we can repost the contents of the archive on a regular basis.
We can even repost all the info of the Wiki so people who don't take the
time look things up can find their answers.
At the very least it would save a fella
Not really. This thread does not belong on this list. It is off topic
and a waste of time for admins dealing w/ real system issues. Like I
said sign up for the qmail list and you will see how a real user list
operates.
LMAO - OK, so those same admins dealing with real system issues are
Hello All,
I remember there is a way to use two Asterisk servers and
allow one to see a virtual trunk that makes it so server B can use the ZAP
channels on server A.
Does anyone know where I can find this? I am racking my
brain trying to remember the terminology.
It was like creating
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Exposing Zap Channels on Server A to be Used ByServer B
On Jun 24, 2005, at 9:07 AM, Wiley Siler
wrote:
Hello All,
I remember there is a way to use two Asterisk servers and
allow one to see
on Server A to be UsedByServer B
On Jun 24, 2005, at 1:31 PM, Wiley Siler
wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear
Sent: Friday, June 24, 2005 9:51 AM
To: Asterisk Users Mailing
List - Non-Commercial Discussion
Subject
Why everyone uses Epia only in mini-itx I still don't get since
horsepower is pretty low for Epia once trans-coding starts...
I would think that Mini-ITX Pentium M is the way to go...
http://www.ibase-i.com.tw/mb890.htm
Throw that in a mini-itx case and you get low power, low wattage high
. If every call that
goes in and out of your Asterisk box is on the same codec and your
voicemail is also recorded on that same codec than Asterisk just
switches the calls like a Pentium 100 linux router switches packets.
On 6/23/05, Wiley Siler [EMAIL PROTECTED] wrote:
Why everyone uses Epia only
LOL - Well, I think we all know a little more about DSL and T1 now at
least
Cheers all,
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nir
Simionovich
Sent: Tuesday, June 14, 2005 6:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Message - From: Wiley Siler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, June 13, 2005 6:59 PMSubject:
[Asterisk-Users] MCI vs. XO/AllegianceHello All,
Anyone out there using ISDN PRI from either MCI or XO/Allegiance?
Gotta make the choice today and the difference
Seems to be all morning. I have not been able to access for several
hours now.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcel van
Kaam, Fonetica
Sent: Tuesday, June 14, 2005 7:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Did they say when it would be corrected?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Sent: Tuesday, June 14, 2005 9:22 AM
To: Matt
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone noticed
Title: Zap Channels
Is there a way to get what channels are not in use from the CLI?
ZAP SHOW CHANNELS just lists the configed channels and ZAP SHOW CHANNEL N just returns OffHook as long as the phone is plugged in.
This is using 2 TDM400 4 port FXO cards ustilizing 6 ports to a channel
Which then presumably leads to higher overselling in the home market
since use is presumed lower.
Also there are often restriction on the line like no Ips given for
servers and no servers allowed.
I doubt they really care if we can afford it persay... I think it is
just a matter of what
because it's handed off as an ethernet -
no need for a csu/dsu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Tuesday, June 14, 2005 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: RE
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Friday, June 10, 2005 7:02 PM
To: Asterisk Users Mailing List -
I have had experience with both the Vina and with XO. If you ask for
it, you should be able to get an Adtran 600 series on the circuit. I
never had any success with the Vina and it really is not a piece of
equipment I would bet the farm on. They may have improved but I would
still just as fo
AdTran can come in either flavor depending on the modules they install.
It can dump analog lines or it can be fully digital and split off voice
T.
I would recommend the digital domain for sure.
Get yourself a Digium T1 card and keep everything digital.
Get a block of DIDs (20 is the norm for
Title: MCI vs. XO/Allegiance
Hello All,
Anyone out there using ISDN PRI from either MCI or XO/Allegiance?
Gotta make the choice today and the difference per month is only about $25 in favor of MCI.
Billing is pretty much the same between the two so I have pretty much no point of
of our
PRI's goes down, we have to notify them almost immediately or they
decommission it so the alert goes away.
Paul
Wiley Siler wrote:
Hello All,
Anyone out there using ISDN PRI from either MCI or XO/Allegiance?
Gotta make the choice today and the difference per month is only about
$25
]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around
:
Wiley
Siler
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, June 13, 2005 6:59 PM
Subject: [Asterisk-Users] MCI vs.
XO/Allegiance
Hello All,
Anyone out there using ISDN PRI from either MCI or
XO/Allegiance? Gotta make the choice
today
] On Behalf Of Wiley
Siler
Sent: Monday, June 13, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Speakeasy SDSL Is 1.5 Megs, is business class (so you get an SLA) and
only costs around $100 per month.
W
Just did the same and it seems (cross fingers) to be fine now too.
However, I have to wonder. What happens to the load on that East Coast
box when we all switch over to it. Sure would be nice to hear from
VoipJet. Considering hwo many times I have recommended them, it would
make me feel better.
LOL
I see. Hmmm... So one could postulate the theory that it is OUR fault
those providers go out of business!
Maybe something like this...
Provider is doing well and giving good service.
Word of mouth increases userbase and service load fo provider.
Provider wants the money obviously and takes
Any other confirmation for this problem? My service seems to be fine
but I have not completed a long duration call yet. I had a user
complain last week about call degradation after 5-10 minutes but that
has been it. I will test some more and let you know. I am on the west
coast server.
Just verified that we are experiencing the same problems. Voice breaks
up when making calls to US destination (I am in PHX, AZ) via the west
coast server.
So far I have not heard back from VoipJet either.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Kind of spawns an interesting side topic though.
I recommend SpamHaus.org for a good blacklist
Easy to integrate into most mail servers and you can't beat free...
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent:
Been that way a couple of hours I think
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Coulthurst
Sent: Thursday, June 09, 2005 2:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] VOIP-INFO
Anyone
Title: AAH 1.1 - CRM Setup
Hello All,
Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH?
I keep getting 'Invalid Channel' but I cannot figure out why.
Thanks!
Wiley
___
Asterisk-Users mailing list
1.1 - CRM Setup
Breifly - yes
in the users extension - define it as SIP/3001 (if the users extension
is 3001)
in the contacts part - define it as you would dial it eg 020 0001 01234
David
On 07/06/05, Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
Has anyone successfully gotten
Do you mean Asterisk @ Home version 0.7 running Asterisk 1.0.7?
The most recent release os AAH is 1.1 just in case you have nto seen the
release notice...
This is something covered repeatedly in the past so the archive will
yeild lots of data.
As I understand it, it is a matter or timing with the
Seshu,
I have Polycom IP500s and I have never had to set that parameter to make
them work with Asterisk. I have used various versions of the BootROM
and sip.ld without any issue.
What problem are you specifically addressing?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
I have IP500s and I use parking via the tranfer method without any
issue.
Callers to my handset are sent to 701 (or other) which is played back.
The call is then dropped on my side.
Do you have details regarding...
What version of * you are using?
Polycom SIP and BootROM levels?
And of course,
Go to dslreports.com and look in the forum for LiveVoip.
Or alternately you can search this list with google via the
site:lists.digium.com parameter.
I spent two months working through problems with LiveVOip.
I highly recommend against them.
Cheers,
Wiley
-Original Message-
From:
and good weekend.
-Scott
- Original Message -
From: Wiley Siler [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 03, 2005 3:36 PM
Subject: RE: [Asterisk-Users] Livevoip 800 Choppy Audio
Go to dslreports.com
With the addition of web based CRM into this machine, is there a plan to
implement traffic shaping in some way?
I know my users will be busy with the CRM side of this but I would not
want it to affect the RTP side fo things if too many users are connected
to the web.
Thanks,
Wiley
Details on IAX trunk can be found here...
http://www.voip-info.org/wiki-IAX
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sandeep
A.S
Sent: Thursday, June 02, 2005 9:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Replacing SIP
You can support as many as you want. You just need to update your
zapata.conf file and change this line...
channel=1-8
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francois
Meehan
Sent: Thursday, June 02, 2005 9:19 AM
To:
This is assuming you have problems with the autoconfig.
The latest seems to add the lines just fine.
When I started using 0.06, I had to do it manually.
W
-Original Message-
From: Wiley Siler
Sent: Thursday, June 02, 2005 9:29 AM
To: '[EMAIL PROTECTED]'; 'Asterisk Users Mailing List
Lance,
Have you configured your sip.conf to use these aprameters under General?
;externip=66.213.227.66
;localnet=192.168.1.0
;localmask=255.255.255.0
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lance
Grover
Sent: Thursday, June 02, 2005 9:39 AM
sig type too fast and shift + enter will send the email early...
OK. Anyway... The parameters below are important for the issue you
have.
The wiki covers this under the sip section www.voip-info.org
W
-Original Message-
From: Wiley Siler
Sent: Thursday, June 02, 2005 9:51 AM
And the point is that this should be a dead issue.
The vendor resolved the problem as quickly as possible and took
responsibility for the mistake.
This really should not be an issue that has to keep going for another
week
W
-Original Message-
From: [EMAIL PROTECTED]
.
W
-Original Message-
From: Francois Meehan [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 02, 2005 10:20 AM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] 2 incoming lines and [EMAIL PROTECTED]
Thanks Wiley,
I was asking
You just need to read up on IAX a little. IAX has no trouble with
firewalling.
As long as the client registers to the IAX server, the path will be
defined and connectivity will occur.
It may look like an odd port if you don't have a static port forward in
place but it will work.
If you really
Are you using custom music files? If so, how did you transfer them to
the box?
If you transferred via FTP, you need to be sure you set the tranfer type
to Binary before sending.
Tranferring using ASCII has always hosed mp3 files for me on the * box.
The net result being similar to your
I would also like to know about compatability of backups and restores.
Does this version restore from any of the other version? Basically 0.9
and higher?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent:
Just a guess if they followed old login/password
It could be...
Maint/password
Or not but worth a try...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Litwiller
Sent: Wednesday, June 01, 2005 2:16 PM
To: Asterisk Users Mailing List -
Links to the firmware can be found from the Wiki.
Check here... www.voip-info.org
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, May 29, 2005 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Well spoken Brian. Karl, let it rest for the sake of Pete.
You made your complaints. It was heard by all.
The facts were laid out about 6 times now.
Everyone has figured out the how and whys of the problem you
experienced.
You have been apologized to and as far as I can see the problem was
Not sure I understand your meaning. You have a phone with 105 as the
registered extension but you want to dial 105 and get voicemail?
Lets assume that *98 is your voicemail extension.
If you dial *98 from any phone it should ask for the extension and
password.
So extension *98 looks something
Well, that will be pretty preferential
As stated before, I love the Polycom IP500. I think it is just a great
phone for less than $200.
It configs easily once you get used to the config file and Polycoms have
great speaker phones.
Many love the Ciscos... Admittedly a beautiful phone but
I thought he meant that as well but I hope that what will occur is that
there is DSL somewhere already that can be utilized.
That conflicts with the 'old town PBX' scenario as well though.
So, assuming there is DSL already, that even makes you wonder why bother
if a phone line already exists and
Wow, I am pretty sure you should update both the bootrom
and sip.ld at the same time.
I would do this with at least one phone and see what
happens.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
CoulthurstSent: Friday, May 27, 2005 11:37 AMTo:
,
Wiley Siler
Who has been drunk with important people
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Nope. I do use their SDSL (rebranded Covad line) for connectivity to
my voip providers though. Works very well.
From what I have seen and heard about their service, it is an all
inclusive package, essentually a hosted VoIP PBX.
Why would you even need * at that point?
W
-Original
Title: Asterisk on 64 bit Linux
Hello All,
After reading a recent exchange off the list archive, I began to wonder if there is any more information on running Asterisk on 64 bit Linux distros.
Of question
1. What performance increase if any?
2. Stability issues?
3. Any issues with the
Did you search for Nortel at www.voip-info.org?
Cheers,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Thursday, May 26, 2005 12:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk connecting to Nortel
News flash. LiveVoip never DID like customers...
I have several nasty emails from them that show how they feel about
customers.
Never threaten this company, we will not tolerate it?
You have got be kidding... Who do they think they are?
What did you do, say you would file with the Better
I officially gave up on LiveVoip yesterday.
I have spoken with Joop (the CTO) on several occasions and he has shined
me a great story of how they are upgrading and all the improvements they
are making. He was always been very personable and good about
explaining what is supposedly going on at
Good questions I have always wondered about since I have a DP box for my
* server.
Isn't that only true in the event that no transcoding is happening?
Won't processor usage increase pretty drastically if transcoding occurs
a lot?
How about if the users have music on hold a lot or use a lot of
Title: Excellent Article explainng what is up with Broadvoice
For those who have had problems of late
http://voxilla.com/voxstory163.html?POSTNUKESID=de6584ef42c2509fb8d0d74403856060
___
Asterisk-Users mailing list
That sounds like a call queue...
http://www.voip-info.org/wiki-Asterisk+call+queues
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Andrews
Sent: Monday, May 23, 2005 10:12 AM
To: Asterisk
Subject: [Asterisk-Users] How do you transfer a call to
SugarCRM looks pretty nice. Are you using the Professional version or
just the open source version?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Monday, May 23, 2005 10:41 AM
To: 'Asterisk Users Mailing List -
.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
|Siler
|Sent: Lunes, 23 de Mayo de 2005 01:22 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] CallerID, TAPI and CTI
|
|SugarCRM looks pretty nice
Title: Junction Networks
Anyone have experience with these guys? If so, good, bad, average?
http://www.junctionnetworks.com
Thanks,
Wiley
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Well, if you are only making office to office calls, save the $500 per
T1 card and just use NICs.
The T1 card is only required if you are using a voice T1. If you are
doing IAX to IAX for example between offices, then Asterisk is your
friend.
Avoid SIP altogether as it is not needed and just use
Yep. T1 card to interface to the old PBX will work.
I thought you were saying you wanted a T1 card to get yoru calls between
offices.
That of course will bemost likelyu accomplished via IAX (across a VPN
channel works well) pver the internet.
If your PBX is included on the Wiki, you should see
Sent: Friday, May 13, 2005 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Other memory stuff
Wiley Siler wrote:
On a similar note, I have a server with 1GB of memory that seems to
never release the memory back to system use.
The system is AAH
Title: Ideal Machine
Hello All,
With the choices of hardware being so varied, I just want to see if anyone can give a recommendation for hardware specific to my needs.
Here are the specs for what we want the new * server to do.
Support some analog phone lines (figure 4-6) via TDM cards
I use AAH with VoipJet and it works perfectly. Setup was a breeze with
absolutely no hand coding of configs required.
VoipJet is without a doubt the best outbound provider I have come
across. No problems at all yet. knock on wood
And the call quality has been awesome.
Anyone having trouble
1.3 cents minute dialing? That is one of the lowest prices out there.
Maybe for you in Australia but in North America, it is a very nice deal.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil
Gupta
Sent: Friday, May 13, 2005 5:49 AM
To: Andrew
It is not that they are not working with Asterisk... It is that there
are none available.
Go check out the link that was sent to you before... Here it is
again
http://www.zapatatelephony.org/
As you can see, Zapata (which drives Asterisk) was originally designed
to be a chipless DSP
Almost positive iLBC is not allowed Use uLaw...
This is directly form the install instructions...
Step 3A (recommended): Set your codec to G.711 ulaw for optimal sound
clarity and minimal transmission delay. In iax.conf (found in
/etc/asterisk) locate the codec section and include the
Good catch. Did not see the FAQ.
Robert, are you the one having problems getting this running in AAH?
W
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 11:22 PM
To: asterisk-users@lists.digium.com; Wiley Siler
Subject: Re: [Asterisk-Users
You are completely correct.
I see by the called number that the user is in Phoenix? I am too.
Call me at 4804230118 ext. 1003 if you want some off list assistance
with this.
I have mine running just fine with AAH 0.09.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Friday, May 13, 2005 10:48 AM
To: Andrew Latham; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] voipjet anyone?
I use AAH with VoipJet and it works perfectly. Setup
Any noticable improvements in performance?
W
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Friday, May 13, 2005 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 64 bit
On Fri, 13
You would want to put the number in the following fields
reg reg.1.displayName=
reg.1.address=
reg.1.label=
reg.1.auth.userId=
Assuming you are using a recent bootrom and sip.
Doesn't the IP500 only support 3 line appearances?
W
-Original Message-
From: [EMAIL PROTECTED]
Title: Other memory stuff
On a similar note, I have a server with 1GB of memory that seems to never release the memory back to system use.
The system is AAH 0.9. Dual AMD Athlon.
This system does IAX out ot my voip providers and has 2 TDM400 cards in it for connection to my POTS lines.
Title: Polycom Bootrom 2.6.2 and SIP 1.5.2
I got em. You want em? Anyone know how I can get these to the site listed on the Wiki?
Thanks,
Wiley
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Title: Polycom Bootrom 2.6.2 and SIP 1.5.2
BTW - Anyone who gets these. Note that the IPMID and
SIP config files are now combined.
Just to save any confusion...
The link from teh Wiki shoudl be updated soon I
think...
Cheers,
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
What bootrom version and sip.ld version are you
using? If you ahve not upgraded, go to teh wiki and get the
latest.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eduardo
JimenezSent: Thursday, May 12, 2005 2:19 PMTo:
asterisk-users@lists.digium.comSubject:
101 - 200 of 432 matches
Mail list logo
