asterisk users - Aug 2004 - No incoming audio on incoming SIP calls

Now this is really frustrating. Everything was working fine, and now it
isn't ... I don't think I've changed anything that would affect
this, but I
guess you never can be too sure.

My setup is as follows:

SIP softphone (SJphone) connected to Asterisk running my Linux NAT firewall
box. This is all on the internal network.

Asterisk then dialing out through various means - SIP to Stanaphone, FWD,
Gossiptel and PSTN via an X100P.

For incoming calls, an 0870 number from CallUK routes to my FWD account, and
an 0870 number from Gossiptel routing to my Gossiptel account.

Outbound calls all work fine ... I get audio in both directions, no problem.

Incoming calls on either 0870 number connect fine, and audio goes from the
softphone to the caller, but not the other way ... I hear no audio on the
softphone from the caller's phone.

I'm getting no alerts from my firewall that it's dropping anything.

I know my way around packet sniffers, but I don't know what to look for
here. What should the inbound audio packets look like?

Thanks


--
David Gurr
Congruity Ltd.
Hemel Hempstead, UK

Hi Dave,

Long time no see...

I have been looking at the various packets going in/out of my network with
regards to my SIP phones.
I usually use the ethereal network sniffer on my network and it has
wonderful support for SIP/RTP/RTSP/SDP analysing.
In the setup phase of a call the RTP packet shows exactly what IPs and
ports that your call is supposed to use.
When I have these problems, I sniff the network and it shows a private IP
instead of the public (external) IP in the voice data stream. Then it's
easy to see what needs to be fixed.
What kind of router/firewall do you use ?

I can send you some examples later on tonight if you need them.

Best regards,
Johan Landerholm, Stockholm, Sweden
(ex. SCO)
> Now this is really frustrating. Everything was working fine, and now it
> isn't ... I don't think I've changed anything that would affect
this, but
> I
> guess you never can be too sure.
>
> My setup is as follows:
>
> SIP softphone (SJphone) connected to Asterisk running my Linux NAT
> firewall
> box. This is all on the internal network.
>
> Asterisk then dialing out through various means - SIP to Stanaphone, FWD,
> Gossiptel and PSTN via an X100P.
>
> For incoming calls, an 0870 number from CallUK routes to my FWD account,
> and
> an 0870 number from Gossiptel routing to my Gossiptel account.
>
> Outbound calls all work fine ... I get audio in both directions, no
> problem.
>
> Incoming calls on either 0870 number connect fine, and audio goes from the
> softphone to the caller, but not the other way ... I hear no audio on the
> softphone from the caller's phone.
>
> I'm getting no alerts from my firewall that it's dropping anything.
>
> I know my way around packet sniffers, but I don't know what to look for
> here. What should the inbound audio packets look like?
>
> Thanks
>
>
> --
> David Gurr
> Congruity Ltd.
> Hemel Hempstead, UK
>
> _______________________________________________
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Solved my own problem ... thought I'd record it here for any others who come
across it.

The problem arises since Asterisk is trying to get out of the way of the
media stream, by doing a SIP re-INVITE to get the two ends of the
conversation to talk directly. This won't work, as Asterisk is telling the
calling party that the IP address to talk to is the private IP address of
the softphone on the internal network. Adding "canreinvite=no" to the
softphone's stanza in sip.conf solves the problem.

It would be helpful if Asterisk noticed that it's about to tell the other
end to use a private IP address ... the ranges are well known, and Asterisk
could do an implicit "canreinvite=no" in this situation.

The same problem didn't occur on outgoing calls as the Dial string includes
a "t" for timeout - as per the wiki, this means that Asterisk must
stay in
the stream to be able to implement this.

Of course, the other way to solve this would be to use a proper SIP proxy
server which handles RTP stream port forwarding ... something I must get
around to.

--
David Gurr
Congruity Ltd.
Hemel Hempstead, UK
> -----Original Message-----
> From: David Gurr [mailto:david.gurr@congruity.co.uk]
> Sent: 04 August 2004 14:05
> To: asterisk-users@lists.digium.com
> Subject: No incoming audio on incoming SIP calls
>
>
> Now this is really frustrating. Everything was working fine, and
> now it isn't ... I don't think I've changed anything that would
> affect this, but I guess you never can be too sure.
>
> My setup is as follows:
>
> SIP softphone (SJphone) connected to Asterisk running my Linux
> NAT firewall box. This is all on the internal network.
>
> Asterisk then dialing out through various means - SIP to
> Stanaphone, FWD, Gossiptel and PSTN via an X100P.
>
> For incoming calls, an 0870 number from CallUK routes to my FWD
> account, and an 0870 number from Gossiptel routing to my
> Gossiptel account.
>
> Outbound calls all work fine ... I get audio in both directions,
> no problem.
>
> Incoming calls on either 0870 number connect fine, and audio goes
> from the softphone to the caller, but not the other way ... I
> hear no audio on the softphone from the caller's phone.
>
> I'm getting no alerts from my firewall that it's dropping anything.
>
> I know my way around packet sniffers, but I don't know what to
> look for here. What should the inbound audio packets look like?
>
> Thanks
>
>
> --
> David Gurr
> Congruity Ltd.
> Hemel Hempstead, UK
>

My Bad, forgot the timeout on the dial string...

But I still cant transfer calls at all. Any ideas?

jd

-----Original Message-----
From: John Howard [mailto:john.howard@adelix.com] 
Sent: 05 August 2004 09:18
To: 'asterisk-users@lists.digium.com'; 'critch@basesys.com'
Subject: RE: [Asterisk-Users] RE: No incoming audio on incoming SIP calls

Hi Steve, 

Sorry to hijack the thread, but I'm confused, when I add 'tr' to the
end of
my dial strings to enable the transferring of that call internally, it
breaks asterisk's dial plan totally.  Calling any extension that has the tr
gives this error:

Aug  4 18:25:33 WARNING[17422]: app_dial.c:838 dial_exec: Invalid timeout
specified: 'tr'

I cant find many resources on this error, do I need to set up the
definitions of t and r (t for 'called extension' transferring, r to tell
the
person the extension is ringing.)

Im using the take16.txt ## transfer patch from twister on the bug fix page
(number 2010) and the 03/08/04 checkout from the cvs.

The ## transfer doesn?t work either needless to say, sending the ## just
goes out over the audio stream as DTMF.

If you can point me at anything in the right direction it would be
appreciated.

Cheers,

jd

-----Original Message-----
t and T are for transfer, not timeout, case denotes which end can
transfer.
-- 
Steven Critchfield <critch@basesys.com>
 

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