asterisk users - Jan 2005 - linphone -> NAT -> * -> NAT -> firefly woes.

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.

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@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@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
>