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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gurr Sent: Wednesday, August 04, 2004 3:56 PM To: [EMAIL PROTECTED] 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:[EMAIL PROTECTED] > Sent: 04 August 2004 14:05 > To: [EMAIL PROTECTED] > 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 > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
