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

[Robinson Tim-W10277]

Sorry, didn't read your mail thoroughly - you've already tried canreinvite=no... 

Next step is to get an Ethereal log from both ends and investigate what is going on with the SIP and RTP packets.

Rgds
Tim
-----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