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Carlos Chavez

2004-Feb-26 17:58 UTC

[Asterisk-Users] GS Budgetone 101 canot receive calls

I just got a Budgetone 101 phone today and after configuring it I can
make calls to any other phone on my * server.  The problem is that no matter
what I do, when I dial the extension assigned to the phone it will always send
me directly to voicemail with the busy message.  

     I tried searching through the mailing list but have not been able to find
a solution.  Can anybody help?  Here is the entry in sip.conf:

[4010]       
username=4010
type=friend
secret=(secret)
host=dynamic
amaflags=default
callerid="Roberto IP Phone" <4010>
mailbox=4010
canreinvite=no 
;reinvite=no
;nat=yes   
qualify=no   
dtmfmode=info
defaultip=192.168.0.102

     I can see on the * console that the phone is registering.  If I do a sip
show peers I ge thw following:

Name/username    Host                 Mask             Port     Status    
4010/4010        192.168.0.102   (D)  255.255.255.255  5060     Unmonitored

     I tried the phone both on the local network and from another network.

--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V.

Matthew B Marlowe

2004-Feb-26 18:08 UTC

head link

[Asterisk-Users] GS Budgetone 101 canot receive calls

Show us your extensions.conf 



Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com

||||
(00)
 ><   Choose a job you love, and you will
/||\  never have to work a day in your life.
=/\
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Carlos
Chavez
Sent: Thursday, February 26, 2004 7:59 PM
To: Asterisk
Subject: [Asterisk-Users] GS Budgetone 101 canot receive calls

     I just got a Budgetone 101 phone today and after configuring it I
can make calls to any other phone on my * server.  The problem is that
no matter what I do, when I dial the extension assigned to the phone it
will always send me directly to voicemail with the busy message.  

     I tried searching through the mailing list but have not been able
to find a solution.  Can anybody help?  Here is the entry in sip.conf:

[4010]       
username=4010
type=friend
secret=(secret)
host=dynamic
amaflags=default
callerid="Roberto IP Phone" <4010>
mailbox=4010
canreinvite=no
;reinvite=no
;nat=yes   
qualify=no   
dtmfmode=info
defaultip=192.168.0.102

     I can see on the * console that the phone is registering.  If I do
a sip show peers I ge thw following:

Name/username    Host                 Mask             Port     Status

4010/4010        192.168.0.102   (D)  255.255.255.255  5060
Unmonitored

     I tried the phone both on the local network and from another
network.

--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V.

_______________________________________________
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

Sergio Serrano Revuelto

2004-Feb-27 01:41 UTC

head link

[Asterisk-Users] GS Budgetone 101 canot receive calls

If your BG 101 is in intranet, try to adjust your qualify parameter to
60.

Regards,

srsergio


-----Mensaje original-----
De: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] En nombre de Matthew B
Marlowe
Enviado el: viernes, 27 de febrero de 2004 2:08
Para: asterisk-users@lists.digium.com
Asunto: RE: [Asterisk-Users] GS Budgetone 101 canot receive calls


Show us your extensions.conf 



Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com

||||
(00)
 ><   Choose a job you love, and you will
/||\  never have to work a day in your life.
=/\
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Carlos
Chavez
Sent: Thursday, February 26, 2004 7:59 PM
To: Asterisk
Subject: [Asterisk-Users] GS Budgetone 101 canot receive calls

     I just got a Budgetone 101 phone today and after configuring it I
can make calls to any other phone on my * server.  The problem is that
no matter what I do, when I dial the extension assigned to the phone it
will always send me directly to voicemail with the busy message.  

     I tried searching through the mailing list but have not been able
to find a solution.  Can anybody help?  Here is the entry in sip.conf:

[4010]       
username=4010
type=friend
secret=(secret)
host=dynamic
amaflags=default
callerid="Roberto IP Phone" <4010>
mailbox=4010
canreinvite=no
;reinvite=no
;nat=yes   
qualify=no   
dtmfmode=info
defaultip=192.168.0.102

     I can see on the * console that the phone is registering.  If I do
a sip show peers I ge thw following:

Name/username    Host                 Mask             Port     Status

4010/4010        192.168.0.102   (D)  255.255.255.255  5060
Unmonitored

     I tried the phone both on the local network and from another
network.

--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V.

_______________________________________________
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 - Feb 2004 - GS Budgetone 101 canot receive calls

[Asterisk-Users] GS Budgetone 101 canot receive calls

[Asterisk-Users] GS Budgetone 101 canot receive calls

[Asterisk-Users] GS Budgetone 101 canot receive calls