[Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Erik Espinoza]

Did you enable passthrough for the rtp ports on the asterisk box?

I had the same problem until I enabled udp 10000:20000 on the firewall.

On Wed, 12 Jan 2005 22:06:18 +0100, Bruno Hertz <brrhtz at yahoo.de> wrote:
> Hi folks
> 
> an issue I don't understand. I'm running * stable 1.0.3 on public
> internet, with following iax.conf / sip.conf entries:
> 
> iax.conf
> 
>  [100]
>  type=friend
>  username=Foo
>  context=default
>  auth=md5,plaintext,rsa
>  secret=secret
>  host=dynamic
>  callerid="Foo" <100>
>  qualify=no
> 
> sip.conf
> 
>  [10]
>  type=friend
>  username=Bar
>  context=default
>  callerid=Bar <10>
>  host=dynamic
>  secret=secret
>  nat=yes
>  canreinvite=no
> 
> On iax exten 10 I register firefly, on sip exten 100 linphone,
> both behind nat.
> 
> Now, calls I can do is e.g.
> firefly -> * -> linphone
> linphone -> * echo test (copied this from demo and put it on exten 600)
> 
> but what wouldn't properly work is is sip to iax bridging
> linphone -> * -> firefly
> 
> More specifically, firefly rings properly, but when I press Accept
> it just keeps ringing, and finally * tells me that linphone didn't
> send any frames:
> 
> channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/10-e8bd
> Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/10-e8bd and IAX2/100/2
> 
> Doing my tcpdumps I checked that there's really no data sent by linphone,
> while nothing is dropped by firewalls either.
> 
> Did anyone experience similar troubles? A hint about how to resolve or further
> debug this would sure be appreciated.
> 
> Another point I'm wondering about is why, in that same connection, the
> caller id handed to firefly is just "10", and not the one specified
> in sip.conf, i.e. "Bar <10>".
> 
> I tested all that stuff also with iaxcomm, i.e. pure iax bridging
> iaxcomm -> NAT -> * -> NAT -> firefly
> and here, everything works OK, calls in both ways and caller id
> transmission.
> 
> Thanks, Bruno.
> 
> _______________________________________________
> 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
>