asterisk users - Mar 2005 - ***SOLVED*** Broadvoice latest changes andstillnot working- An Additional Server****Solved*****!

Van,

It's a new version and there is no inventory in stores yet

I know you'll be pleased once it finally arrives.

Best,

William


-----Original Message-----
From: Zanzamar Majere <Phoneman@wbtllc.com>
Date: Thu, 10 Mar 2005 11:05:07 
To:Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and
	stillnot	working- An Additional Server
	****Solved*****!

I had this same problem.  It was still not authenticating on Invite when you 
look at it in non sip debug mode, just asterisk -vvvvvvvvr

I found that using just the generic sip.broadvoice.com and making sure I had 
the right settings in my Sip.conf worked for me

On Thursday 10 March 2005 10:37 am, Joe wrote:
> Mark and all,
>
>  I rebuilt as well,  but I do not have the same results as you.  400 bad
> response and 401 not authorized.  I dial out just fine,  but as soon as
> the phone answerers on the other end,  I start getting 401 not
> authorized and 400 bad response errors and no audio is sent.  I can hear
> the phone picked up and audio for 1/4 of a second,  and then it goes
> silent.  I hear a clicking sound when I send outbound calls and pick up
> the receiving party phone.  I'm using the default
> proxy.dca.broadvoice.com server.
>
> Joe
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of MF Hulber
> Sent: Wednesday, March 09, 2005 8:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and
> stillnot working- An Additional Server ****Solved*****!
>
> I concur.  I rebuilt today and now I seem to be able to dial out.
>
> MARK.
>
> Chris Nibeck wrote:
> > thank you everyone!
> >
> > It does not seen that it was configuration problems at all.
> >
> > It appears it was the CVS that I was using from yesterday.
> >
> > I decided to start over, downloaded the latest CVS, recompiled, and
> > voila!  * started working!!!!
> >
> > Indeed even a Cisco ATA that was never working before started working!
> >
> >
> > Thanks to everyone!
> >
> > Chris
> >
> > On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote:
> >> there are two of us with the same problem so I will answer for me.
> >> Yes I tried the below instructions.
> >>
> >> The current thinking by multiple people is * never tries
> >> authenticating so removing the FQDN will force * to go to the
related
> >>
> >> section named by either a phone number or a non Fully Qualified
> >> Domain Name.
> >>
> >> But I still don't have it working so who knows.
> >>
> >> Anyone that wishes to call me via BV my number is 8475100139 and
it
> >> is up.
> >>
> >> Chris
> >>
> >> On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote:
> >>> Have you tried this:
> >>>
> >>>
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
> >>>
> >>> Zanzamar Majere wrote:
> >>>> Thank you for the response.   I still have the errors
mentioned
> >>>> below, sip response and Failed to authenticate on INVITE
> >>>>
> >>>> [PPPPPPPPPP]
> >>>> type=peer
> >>>> username=PPPPPPPPPP
> >>>> fromuser=PPPPPPPPPP
> >>>> authuser=PPPPPPPPPP
> >>>> fromdomain=sip.broadvoice.com
> >>>> secret=XXXXXXXXXX
> >>>> host=sip.broadvoice.com
> >>>> dtmfmode=inband
> >>>> insecure=very
> >>>> context=sip
> >>>> qualify=yes
> >>>> disallow=all
> >>>> allow=ulaw
> >>>> allow=gsm
> >>>> ;Disable canreinvite if you are behind a NAT
> >>>> ;canreinvite=no
> >>>> nat=no
> >>>>
> >>>> Does anyone else have any other suggestions?
> >>>
> >>> _______________________________________________
> >>> 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
>
>
> _______________________________________________
> 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
_______________________________________________
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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

Sent from my BlackBerry - please excuse any typos.

Zanzamar, 

 I agree that it should work.  I can call out and have the land phone
ring,  but as soon as it is answered, another invite goes out and that
is when I get the 401 not authorized. I don't want to go down this
route,  but could this be a Codec issue?

Here is my sip config



[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser= BBBBBBBBBB
username= BBBBBBBBBB
authuser= BBBBBBBBBB
secret= secret
context=sip
nat=no
insecure=very
dtmfmode=inband




-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of William
Sent: Thursday, March 10, 2005 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest
changesandstillnot working- An Additional Server****Solved*****!

Van,

It's a new version and there is no inventory in stores yet

I know you'll be pleased once it finally arrives.

Best,

William


-----Original Message-----
From: Zanzamar Majere <Phoneman@wbtllc.com>
Date: Thu, 10 Mar 2005 11:05:07 
To:Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and
	stillnot	working- An Additional Server
	****Solved*****!

I had this same problem.  It was still not authenticating on Invite when
you 
look at it in non sip debug mode, just asterisk -vvvvvvvvr

I found that using just the generic sip.broadvoice.com and making sure I
had 
the right settings in my Sip.conf worked for me

On Thursday 10 March 2005 10:37 am, Joe wrote:
> Mark and all,
>
>  I rebuilt as well,  but I do not have the same results as you.  400
bad
> response and 401 not authorized.  I dial out just fine,  but as soon
as
> the phone answerers on the other end,  I start getting 401 not
> authorized and 400 bad response errors and no audio is sent.  I can
hear
> the phone picked up and audio for 1/4 of a second,  and then it goes
> silent.  I hear a clicking sound when I send outbound calls and pick
up
> the receiving party phone.  I'm using the default
> proxy.dca.broadvoice.com server.
>
> Joe
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of MF
Hulber
> Sent: Wednesday, March 09, 2005 8:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes
and
> stillnot working- An Additional Server ****Solved*****!
>
> I concur.  I rebuilt today and now I seem to be able to dial out.
>
> MARK.
>
> Chris Nibeck wrote:
> > thank you everyone!
> >
> > It does not seen that it was configuration problems at all.
> >
> > It appears it was the CVS that I was using from yesterday.
> >
> > I decided to start over, downloaded the latest CVS, recompiled, and
> > voila!  * started working!!!!
> >
> > Indeed even a Cisco ATA that was never working before started
working!
> >
> >
> > Thanks to everyone!
> >
> > Chris
> >
> > On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote:
> >> there are two of us with the same problem so I will answer for me.
> >> Yes I tried the below instructions.
> >>
> >> The current thinking by multiple people is * never tries
> >> authenticating so removing the FQDN will force * to go to the
related
> >>
> >> section named by either a phone number or a non Fully Qualified
> >> Domain Name.
> >>
> >> But I still don't have it working so who knows.
> >>
> >> Anyone that wishes to call me via BV my number is 8475100139 and
it
> >> is up.
> >>
> >> Chris
> >>
> >> On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote:
> >>> Have you tried this:
> >>>
> >>>
http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup
> >>>
> >>> Zanzamar Majere wrote:
> >>>> Thank you for the response.   I still have the errors
mentioned
> >>>> below, sip response and Failed to authenticate on INVITE
> >>>>
> >>>> [PPPPPPPPPP]
> >>>> type=peer
> >>>> username=PPPPPPPPPP
> >>>> fromuser=PPPPPPPPPP
> >>>> authuser=PPPPPPPPPP
> >>>> fromdomain=sip.broadvoice.com
> >>>> secret=XXXXXXXXXX
> >>>> host=sip.broadvoice.com
> >>>> dtmfmode=inband
> >>>> insecure=very
> >>>> context=sip
> >>>> qualify=yes
> >>>> disallow=all
> >>>> allow=ulaw
> >>>> allow=gsm
> >>>> ;Disable canreinvite if you are behind a NAT
> >>>> ;canreinvite=no
> >>>> nat=no
> >>>>
> >>>> Does anyone else have any other suggestions?
> >>>
> >>> _______________________________________________
> >>> 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
>
>
> _______________________________________________
> 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

Sent from my BlackBerry - please excuse any typos.
_______________________________________________
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