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@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${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
******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 => _1NXXNXXXXXX,1,Dial,IAX2/xxxxx@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN}
*****IAX.CONF*****
[general]
port=5036
bind=0.0.0.0
bandwidth=low
disallow=lpc10
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
disallow=all
allow=ilbc
allow=gsm
allow=ulaw
******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
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@radform.com>
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: <200503130823.j2D8NdUw012658@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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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" <edward@radform.com>
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <200503111016.j2BAFw1s014610@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@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${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