[Asterisk-Users] NuFone Configuration [problem]

[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" <edward at radform.com>
Subject: RE: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue
	88
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <200503130823.j2D8NdUw012658 at wrench.thebook.com>
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeff
Glassman
Sent: Sunday, March 13, 2005 12:17 AM
To: asterisk-users at 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" <edward at radform.com>
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
	<asterisk-users at lists.digium.com>
Message-ID: <200503111016.j2BAFw1s014610 at wrench.thebook.com>
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=xxxxxx

***extensions.conf:***

exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx at NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx at NuFone/${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--> ASTERISK----IAX--->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