[Asterisk-Users] DID in 513 Cincinnati

2005-06-26 Thread Jeff Glassman 
Message: 19
Date: Sun, 26 Jun 2005 12:12:46 -0400
From: John Kington [EMAIL PROTECTED]
Subject: [Asterisk-Users] 
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii; format=flowed

Does anyone have a recommendation for a DID local to Cincinnati (513)? I
am
looking for a pay as you go solution for incoming calls with light
usage. I 
would
prefer IAX but can use SIP solution.
Regards,
John


Try telasip.com for SIP I am using them in 614 (Columbus) or
www.teliax.com, I have not used them but have heard good things.

Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] Changing Caller ID

2005-06-26 Thread Jeff Glassman 








I have two X100P clone cards working perfectly
in my asterisk box, these lines are off an analog extension from a PRI.



They each have DID # assigned to them and I can call the DID
and receive calls. When I make an
outgoing call using the Zap trunk the caller ID is of the PRI line. Is there any way to change the caller ID
to the DID assigned to the line?



Thanks in advance,



Jeff






___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] WRT54GP2A-AT

2005-05-13 Thread Jeff Glassman 








My Wireless router needs replacing, I have seen a wirerless/router/ata
that is locked into A TT service, http://www.linksys.com/products/product.asp?prid=662scid=35



Can theses be unlocked or can you buy them without being
tied in to a VOIP provider



Jeff






___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] unlimited iax termination

2005-04-09 Thread Jeff Glassman 
Message: 11
Date: Sat, 9 Apr 2005 08:21:16 -0700
From: Kerry Garrison [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] unlimited iax termination
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;   charset=us-ascii

I am trying to put together a matrix. Please send me links, corrections,
additions, flames, etc.

http://www.geekgazette.com/index.php?option=com_contenttask=viewid=25;
Item
id=26

-Kerry

Great idea, I would like to see one for DID/800 incoming also.

Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 67

2005-04-07 Thread Jeff Glassman 

Message: 6
Date: Thu, 7 Apr 2005 16:24:18 -0700
From: snacktime [EMAIL PROTECTED]
Subject: [Asterisk-Users] Getting a good deal on a PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

We have 10 incoming POTS lines to our offices, and a nortel norstar pbx.
I've been looking at replacing it with * at some point in the future,
and the point that looks most cost effective is when we move to PRI.

Problem is, I'm not really sure how to go about getting a good deal, or
what questions to ask.  90% of calls will be inbound.  I called up Qwest
and they quoted me $800 month.  I haven't called up any CLEC's yet to
see what they can do.

Any suggestions?  We are in Seattle, Washington.

Chris


In Columbus Ohio we pay about $600.00 per month for a PRI from Time
Warner. Unlimited incoming/outgoing.  


Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 67

2005-04-07 Thread Jeff Glassman 

Message: 6
Date: Thu, 7 Apr 2005 16:24:18 -0700
From: snacktime [EMAIL PROTECTED]
Subject: [Asterisk-Users] Getting a good deal on a PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

We have 10 incoming POTS lines to our offices, and a nortel norstar pbx.
I've been looking at replacing it with * at some point in the future,
and the point that looks most cost effective is when we move to PRI.

Problem is, I'm not really sure how to go about getting a good deal, or
what questions to ask.  90% of calls will be inbound.  I called up Qwest
and they quoted me $800 month.  I haven't called up any CLEC's yet to
see what they can do.

Any suggestions?  We are in Seattle, Washington.

Chris


In Columbus Ohio we pay about $600.00 per month for a PRI from TW
Telecom. Unlimited incoming/outgoing.  


Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] trying to add the free voipjet test to my

2005-03-27 Thread Jeff Glassman 
Message: 10
Date: Sat, 26 Mar 2005 23:19:17 -0500
From: Jon Walsh [EMAIL PROTECTED]
Subject: [Asterisk-Users] trying to add the free voipjet test to my
asteriskat home???
To: Asterisk-Users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1

No Dice so far,  anyone now how to add anIAX trunk? What are the
settings exactly? I have added everything but I do not know what are the
registration strings? Jonathan

-

I set it up manually

iax.conf

[voipjet]
type=peer
secret=PW
notransfer=yes
host=216.118.117.46
context=default
auth=md5
   

extensions.conf

exten = _1NXXNXX,1,SetCallerID(1234567890); Set your CallerID as a
ten digit number like this. See our FAQ 
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com
NANPA 
exten = _011.,1,SetCallerID(1234567890); Set your CallerID as a ten
digit number like this. See our FAQ. 
exten = _011.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} ; VoipJet.com WORLD 
;Do not change IAX2/2082 in the above two lines!

Replace (1234567890) with your own caller ID

Replace the 1234 in [EMAIL PROTECTED] with your account number  

However it will use voipjet as the default trunk when you
dial 1 xxx xxx 

I have not been able to put a prefix dial out for it


Jeff



___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] Backup for linux/asterisk

2005-03-24 Thread Jeff Glassman 








After getting my feet wet with [EMAIL PROTECTED], I want to set up
a second asterisk box to add a call shop billing and other add-ons such as LCR.



My question is as follows. Is there a backup program that will save
to a tape drive or a USB CD Writer so if I mess up an install I dont
have to go through a complete reinstall?
I saw a few programs out
there but they required X windows and from what I read it is suggested that X
windows not be installed on an Asterisk box.



Thanks,



Jeff






___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] RE:Newbie question

2005-03-20 Thread Jeff Glassman 
Message: 5
Date: Sun, 20 Mar 2005 03:55:50 +0100
From: bram [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE:Newbie question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain

It said 'include zapata-channels.conf', where this line wasn't
commented bij the ';'...

Could you post me a working example of such a config (or a part of it,
for the X100P cards...?

Thanks guys!


Here is my Zapata,conf

Do not worry about zapata-channels.conf' It is generated automaticly DO
NOT CHANGE IT  All I changed was

; channel = 1  to this channel = 1

Jeff


;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

channel = 1 
#include zapata-channels.conf


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] newbie question

2005-03-19 Thread Jeff Glassman 
bram kortleven Wrote

Message: 6
Date: Sat, 19 Mar 2005 22:16:39 +0100
From: bram kortleven [EMAIL PROTECTED]
Subject: [Asterisk-Users] newbie question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain

I guess the first time it didn't get through... I didn't see it appear
in the list, that is...


I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts
on it. They seem to run fine inside my network, so that's OK. Now, I
want to start using a X100P to connect it to my phone line, to make
call routing between internal SIP phones/softphones, my local phoneline
and an external SIP server. How do I enable and configure the X100P?

I ran the configuration tool locally on the machine (the genzaptelconf
thing) and it added a line to the config.
Now using the number it gave me, in the trunk config in AMP, I still
cannot get an outside line (connected it to a simple analogue pbx
system) and call outside the *-server..
Could anyone help me with this?
Thanks guys

You need to go into the Zapta.conf and remove the semi colon

; channel = 1

Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] Backing up configurations and *@home list?

2005-03-17 Thread Jeff Glassman 














Date: Thu, 17 Mar 2005 14:06:48 -0800
From: Don Murray [EMAIL PROTECTED]
Subject: [Asterisk-Users] Backing up configurations and [EMAIL PROTECTED] list?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed


Hello again,

I have [EMAIL PROTECTED] set up and working.  There are a couple of questions I 
have about system administration that I couldn't find on the wiki 
(although I may not be using the right buzzwords).

(1) is there an [EMAIL PROTECTED] specific mailing list?  I believe someone 
on this list mentioned there was but I cannot find it at sourceforge, 
the [EMAIL PROTECTED] web page, or the asterisk.org web page.


 (2) is there a utility for backing up asterisk configurations and 
current status?  I'd like to be able to do regular back-ups and if a big

problem happens, have a method to easily re-install the system on a 
different machine.  Is there an *-friendly way of doing this or should I

just do directory dumps periodically and copy them back into place after

a fresh install?

The built in AMP interface takes care of backup and restore

Thanks

Don

[EMAIL PROTECTED] info

https://sourceforge.net/forum/?group_id=123387

http://www.techdatapros.com/asterisk/


Backup and restore info

The built in AMP interface takes care of backup and restore


Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk@Home

2005-03-14 Thread Jeff Glassman 
Scheda [EMAIL PROTECTED] wrote:
 Have any of you tried this?
 
 http://asteriskathome.sourceforge.net/
 
 I'm thinking of using this version. I'm debating
 between it and
 Knoppix with Asterisk thrown in there as well. I'm a
 linux newbie for
 the most part, but can get around and get done what
 I need done with
 help here and there, but I don't know if [EMAIL PROTECTED] is
 all what I need.
 
 Here is what I need * to do for me pretty much.
 
 -Voicemail
 -Conferencing
 -IM callbacks (Instant message from my cell and it
 calls me back)
 -Extentions
 -A few other things which I can handle
 
 I've used * a tad bit in the past, no real heavy
 work with it though.
 I would think that the GUI in [EMAIL PROTECTED] would make
 Asterisk less
 functional. Seeing as how I'm not using this for
 some large business
 or anything, just as something so listeners of a
 radio show I do can
 interact, I don't think [EMAIL PROTECTED] would be WAY too
 restrictive on what I
 need accomplished.
 
 If you have used this before, can you post a review
 on what you think
 of it or just tell me if this will suit my needs or
 not?

I tried both and [EMAIL PROTECTED] is so much easier to use and learn.  With
AMP built into [EMAIL PROTECTED] configs are a snap.  Plus its has its own
mailing list.

Jeff


___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] NuFone Configuration [problem]

2005-03-13 Thread Jeff Glassman 
Someone once said YOU CAN'T BE TO RICH OT HAVE TOO MUCH BANDWITH

1 How much do you have?  How many phone calls and how many other users
on your connection?

2 Go to http://testmyvoip.com/ and test your bandwith


Jeff

Date: Sun, 13 Mar 2005 09:23:35 +0100
From: Edward Banfa [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue
88
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;   charset=US-ASCII

Hi,
Thanks for the reply. I tried changing my allow and disallow entries to
match yours below but still no luck.
Could my problems be bandwidth related? If so what amount of bandwidth
should I request?

Cheers 

Edward


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Glassman
Sent: Sunday, March 13, 2005 12:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88

These allow and disallow work with NuFone for me


disallow=all
allow=ulaw
allow=alaw
allow=gsm

Jeff

Message: 11
Date: Fri, 11 Mar 2005 11:15:51 +0100
From: Edward Banfa [EMAIL PROTECTED]
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;   charset=us-ascii



Hello,
I am trying to configure the my asterisk box here with the following

**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx

***extensions.conf:***

exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}

I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan. The mediatrix talks sip to the
asterisk
box on the lan. We are running asterisk  on FC3 .

SOFTPHONES[XLITE] ---SIP-- ASTERISKIAX---NUFONE[ASTERISK]

ANALOGPHONES---MEDIATRIX_1102---SIP---ASTERISK---IAX---NUFONE[ASTERISK
]

Well the problem goes something like this.
1) I can dial a number form the softphones and when the call is answered
I
can hear the user on the other end but the user can't hear me
2) I can dial a number from the analog phones (via mediatrix tru to
asterisk)(the mediatrix is properly registered with our asterisk box)
and
when the call is answered both ends can't hear a word, its just silent.

I think I am having a codec problem here. What am I doing wrong. We
would
sincerely appreciate any help/pointers.

Thank you all
Edward Banfa



___
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/mailman/listinfo/asterisk-users

[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 88

2005-03-12 Thread Jeff Glassman 
These allow and disallow work with NuFone for me


disallow=all
allow=ulaw
allow=alaw
allow=gsm

Jeff

Message: 11
Date: Fri, 11 Mar 2005 11:15:51 +0100
From: Edward Banfa [EMAIL PROTECTED]
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;   charset=us-ascii



Hello,
I am trying to configure the my asterisk box here with the following

**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx

***extensions.conf:***

exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}

I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan. The mediatrix talks sip to the
asterisk
box on the lan. We are running asterisk  on FC3 .

SOFTPHONES[XLITE] ---SIP-- ASTERISKIAX---NUFONE[ASTERISK]

ANALOGPHONES---MEDIATRIX_1102---SIP---ASTERISK---IAX---NUFONE[ASTERISK
]

Well the problem goes something like this.
1) I can dial a number form the softphones and when the call is answered
I
can hear the user on the other end but the user can't hear me
2) I can dial a number from the analog phones (via mediatrix tru to
asterisk)(the mediatrix is properly registered with our asterisk box)
and
when the call is answered both ends can't hear a word, its just silent.

I think I am having a codec problem here. What am I doing wrong. We
would
sincerely appreciate any help/pointers.

Thank you all
Edward Banfa

**EXTENSION.CONF***
[general]
static=yes

[from-sip]
exten = 100,1,Dial(SIP/edward,20)
exten = 100,2,Hangup

exten = 101,1,Dial(SIP/phone1,20)
exten = 101,2,Hangup

exten = 102,1,Dial(SIP/phone2,20)
exten = 102,2,Hangup

exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}


*IAX.CONF*
[general]
port=5036
bind=0.0.0.0
bandwidth=low
disallow=lpc10

[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx
disallow=all
allow=ilbc
allow=gsm
allow=ulaw


disallow=all
allow=ulaw
allow=alaw
allow=gsm


**SIP.CONF*
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[edward] ;My Xlite softphone
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=edward 100
mailbox=100
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726

[phone1] ;First analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone1 101
mailbox=101
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726

[phone2] ;Second analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone2 102
mailbox=102
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726














--

Message: 12
Date: Fri, 11 Mar 2005 15:57:38 +0530
From: Jagan Mohan [EMAIL PROTECTED]
Subject: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using
SIP load balancer
To: Asterisk asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII

Hi,

  I'm trying to do load balancing between 2 asterisk servers using SIP 
load balancer, provided by http://www.vovida.org

  I used the following options on lbproxy, but I get the below message 
continuously. 

./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2

No proxies are up - can not send message to anyone

Xlite is not able to register to the asterisk server.

Is there anything which needs to be tweaked on Asterisk side to get this
working? Please help.

Thanks,
Jagan


--

Message: 13
Date: Fri, 11 Mar 2005 11:31:29 +0100
From: Vledder, Hans [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:

[EMAIL PROTECTED]
m

Content-Type: text/plain;   charset=iso-8859-1

Hi Steve,

Since you don't mention what USB ISDN adapter specifically you are
thinking about, what do you think we will be able to tell you.

All I know about the adapter is what I've told you. It's a USB
Colognechip
based ISDN controller - probably HCF-USB based. It's supported by Linux,
but
there's no info on access to B and D channels.

Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...


On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote:
 Guys,
 
 I am planning on building a small SIP PBX with a single ISDN line.
Currently
 I am looking into the specs of a very tiny barebone system that has an
 option Colognechip base ISDN

RE: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-01-25 Thread Jeff Glassman 
They are coming out with a patch for the DID problem tonight.  Need to have
Asterisk 1.0.3

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tim Lewis
Sent: Tuesday, January 25, 2005 6:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Anyone having problems with LiveVoIP?


I am having two problems. The first one is about half the time asterisk
fails to read the DTMF tones. The second is with my 3 DID's some times
it goes through and other times it does now.  Right now it does nothing.
Sometimes it rings for ever. With no out put on the asterisk console.
They don't like to answer the phone or respond to email's is a timely
matter.

Anyone else having these issues?


-Thanks

___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-01-25 Thread Jeff Glassman 
LiveVoip has developed a patch for a problem that our upstream gateway to
Level 3 is having with Asterisk users. We hope to deploy this patch near the
evening and are sorry for the delay. This problem stems from codec issues in
Asterisk. You must use Asterisk ver. 1.0.3 as well. As soon as our engineers
have tested this on a development platform the production equipment will be
upgraded. We need to do this after 7 P.M. EST which is when high volume
calls drop off. There are additional DTMF issues people are reporting, that
again are Asterisk based problems. Our in-house team continue to evaluate
these as well. This may or may not be an issue in your case.



Network Operations Team
LiveVoip LLC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tim Lewis
Sent: Tuesday, January 25, 2005 9:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Anyone having problems with LiveVoIP?


Thanks Jeff!

I think it's a little too late to find this info out. 3 to 4 days of no
service. I have send many emails and still awaiting a response. Reminds
me of my ILEC (QWEST)

Do you have any info on what this patch does?


-later

On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote:
 They are coming out with a patch for the DID problem tonight.  Need to
have
 Asterisk 1.0.3

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Tim Lewis
 Sent: Tuesday, January 25, 2005 6:33 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Anyone having problems with LiveVoIP?


 I am having two problems. The first one is about half the time asterisk
 fails to read the DTMF tones. The second is with my 3 DID's some times
 it goes through and other times it does now.  Right now it does nothing.
 Sometimes it rings for ever. With no out put on the asterisk console.
 They don't like to answer the phone or respond to email's is a timely
 matter.

 Anyone else having these issues?


 -Thanks

 ___
 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/mailman/listinfo/asterisk-users

 ___
 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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Anyone having problems with LiveVoIP?

2005-01-25 Thread Jeff Glassman 
Does anyone know if [EMAIL PROTECTED], which is running Asterisk
CVS-v1-0-01/18/05-11:35:19,

can be upgraded to 1.0.3 or higher?

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Dingman
Sent: Tuesday, January 25, 2005 9:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anyone having problems with LiveVoIP?


LiveVoip has a problem with Asterisk users on versions less than 1.0.3  If
you are not using that version you need to upgrade now.
We have a problem with two of our carriers at their gateway related to the
Asterisk users. Our staff has developed a patch that is
being tested at this time. Once the patch has been approved on our testbed
we will move it on to the production switch environment.
We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due
to high traffic loads. We expect to do switch updates after
7 p.m. this evening that should resolve the problems you are having.

LiveVoip engineers are also looking at a DTMF problem in the Asterisk
software ver. 1.0.3 which may or may not involve you. Both of
these issues are Asterisk software related in nature and not LiveVoip LLC
switching defects.

Thank You in Advance for your understanding. This issue has been placed
under a master ticket for tracking.

** When contacting LiveVoip LLC Support please provide us with the latest
version of Asterisk you are using, any and all logs if
necessary and as much detail regarding any problems you are having.

Network Operations Team
LiveVoip LLC


On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis [EMAIL PROTECTED] wrote:
 Thanks Jeff!

 I think it's a little too late to find this info out. 3 to 4 days of no
 service. I have send many emails and still awaiting a response. Reminds
 me of my ILEC (QWEST)

 Do you have any info on what this patch does?

 -later

 On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote:
  They are coming out with a patch for the DID problem tonight.  Need to
have
  Asterisk 1.0.3
 
___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net - GREATadvance]

2005-01-24 Thread Jeff Glassman 
I agree that using a home phone could be a pain in the middle of the night
but I have an available POTS line off my T1 so I made it available for local
calls to the 614 area code!


Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Duane
Sent: Monday, January 24, 2005 10:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial
and Business-Oriented Asterisk Discussion
Subject: Re: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net -
GREATadvance]


Duane wrote:

I was discussing bellster with a friend of mine, and he made another
point about this service...

 I can't imagine how unsettling it would be for my girlfriend to pick up
the phone and hear somebody else on the line.  The first time that happened,
that'd be the end of me sharing the line with anybody.  I can't think of a
way that I would explain Bellster to a non-geek in a way that makes them
comfortable.

 I can almost hear her saying, How much do you make a year again?  And how
much is unlimited long distance from the phone company?  Is that going to
break you financially?  Because if it is, you've either got a drug problem,
gambling problem, or a hooker problem.

Made me laugh at least :)

Another small point is that a lot of countries don't have flat rate
calls, and I highly doubt anyone in those countries would be offering
their land lines for this kind of service either. It costs me between 20
and 30c per call to make local calls, so this basically only leaves
North American and New Zealand as the only viable options that I know of.

--

Best regards,
  Duane

http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers

I do not try to dance better than anyone else.
 I only try to dance better than myself.
___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?

2005-01-12 Thread Jeff Glassman 
It is a USB attached phone

It needs to be used with a soft phone

It does work as a handset for x-ten type soft phone or their own softphone 


Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kim Lux
Sent: Wednesday, January 12, 2005 9:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone
?


http://www.pcphoneline.com/skype


The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support.  Skype is not
forecast to have SkypeIn available until June 2005 but you can have
the capability now via its built in SIP capabilities.

Is this a wireless USB phone ?  Does it support SIP and could it be used
to connect to any SIP server ? 

Does anyone have experience with these ?

Thanks. 

-- 
Kim Lux,  Diesel Research Inc.


___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Xorcom Rapid CD for Production?

2005-01-03 Thread Jeff Glassman 
When I tried to use Rapid CD I was able to get it loaded but never was able
to get get my internet working either by dcph or by assigning the ip
address.

I havve tried two other flavors of Linux and both connected to the internet.

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Monday, January 03, 2005 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Xorcom Rapid CD for Production?


Hi All,

The past day or so I've setup a new * server based upon the Xorcom
Rapid ISO. It did as promised; wiped the base system, installed Debian
OS, installed Asterisk with a dummy configuration. So far so good.

If I could get the config from my existing * server migrated to the
Xorcom box the I'd be ready to roll. Essentially, it would be the same
as I have now on Fedora Core 1, but with a text mode management shell
and reboots into everything fully working without an user intervention.
I could put a 1 Gb CF card in a CF to IDE adapter and have an HD free
server, or just use a small HD instead.

However, I can't seem to get my current * configs to load to the Xorcom
box. I tried to sftp from my Windows desktop but my Windows ssh/sftp
client (www.privateshell.com) would not connect to the Xorcom box. I
confirmed that sshd was installed and running by trying Putty, as
described in the Xorcom faq. With Putty and its sftp counterpart I
could one-by-one upload the conf files from my existing server to the
Xorcom box, but it killed the * install on the Xorcom box.

Anybody have any experience turning the Xorcom Rapid installation into
a production installation for a small office? It would seem that I only
need to setup the configs appropriately, but something is going
dramatically wrong and I can't see it at the moment.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] is wiki drunk

2005-01-01 Thread Jeff Glassman 
Must be the whole site is down

Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed
Sent: Friday, December 31, 2004 6:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] is wiki drunk


is there any problem with wiki
___
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/mailman/listinfo/asterisk-users

___
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/mailman/listinfo/asterisk-users