[Asterisk-Users] Re: Phones can talk to asterisk but not each other through it

[Robinson Tim-W10277]

Try setting 'reinvite=no' in the sip.conf file.  This will force Asterisk to stay in the loop...it otherwise tries to step out of the connection and let the phones talk directly to each other,which is fine on a LAN but if both are behind NAT firewalls is asking for complication.


Rgds
Tim Robinson
Basingstoke, UK

-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of tony at softins.clara.co.uk
Sent: 24 March 2004 13:31
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Phones can talk to asterisk but not each other through it


In article <40618664.bb.fa8.14813 at yponeinc.com>,
 <asterisk-users at lists.digium.com> wrote:
> Tony,
> What is the BW connectivity at the [*] box?

It's lots. Much more than the broadband connections our phones are behind. File downloads to the * box from elsewhere on the internet typically go at several hundred kbytes/sec.

> You may try to set the GS phones to GSM codec to reduce BW, and see if 
> that improves the situation.

I didn't have much success using GSM, but I'll try again.

Thanks for the reply...

Tony

> WW
> ----- Original Message Follows -----
> > I posted this a week or two ago but no replies, so trying again...
> > 
> > Summary: Two phones in different locations, each behind NAT, can 
> > both talk to an Asterisk server on the net, for the demo or for 
> > voicemail, but can't maintain a call to each other via that 
> > asterisk.
> > 
> > Original post with details:
> > 
> > I have a problem with an installation of asterisk on my colo server. 
> > I have a Grandstream BT102 behind a Linux NAT firewall, and my 
> > colleague also has one behind his.
> > 
> > My connection is ADSL with 512k down and 256k up. My colleague's is 
> > Cable with 600k down and I don't know whether it's 128k or 256k up.
> > 
> > I have the phones set up in sip.conf with nat=yes, qualify=yes and 
> > canreinvite=no. Each phone can successfully connect with Asterisk 
> > and dial the Asterisk Demo, leave and pick up voicemail, etc.
> > 
> > However, if one phone tries to dial the other, once the called phone 
> > is answered, the audio starts off very stuttery and broken, and 
> > after a few seconds dies completely and the call gets dropped.
> > 
> > In the asterisk log there are many entries for that time
> > saying: Recv error: Resource temporarily unavailable.
> > 
> > I am using the zaprtc timer module on the asterisk server, but in 
> > any case I understood that was only required for MeetMe or MOH.
> > 
> > The server system is a Duron XP 1800, with 512MB RAM, running Fedora 
> > Core 1 with updates, and a standard 2.4.22 kernel that was 
> > recompiled only to make the RTC a module instead of compiled in (so 
> > I could rmmod it and then load zaprtc instead, which works fine).
> > 
> > Can anyone suggest what things I should check or change?
> > 
> > Cheers
> > Tony
> > --
> > Tony Mountifield
> > Work: tony at softins.co.uk - http://www.softins.co.uk
> > Play: tony at mountifield.org - http://tony.mountifield.org
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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
> 
> Willy Wouters
> ypOne Publishing
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
> 


-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org _______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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