[Asterisk-Users] RE: No incoming audio on incoming SIP calls

[Luke Catranis]

I must do the same with the proxy... one note... the t stands for transfer
per the wiki: t : Allow the called user to transfer the call

http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial



-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of David Gurr
Sent: Wednesday, August 04, 2004 3:56 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] RE: No incoming audio on incoming SIP calls

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 at congruity.co.uk]
> Sent: 04 August 2004 14:05
> To: asterisk-users at 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
>

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