Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok).
In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:[EMAIL PROTECTED] f:"Test User" <sip:[my phone [EMAIL PROTECTED]>;tag=as341d210b t:<sip:[EMAIL PROTECTED]>;tag=b6e96dae-1dd1-11b2-a01e-b03162323164+b6e9 6dae m:sip:[EMAIL PROTECTED]:5075 c:application/sdp l:170 v=0 o=- 3459442714 3459442714 IN IP4 192.168.201.25 s=SIP Call c=IN IP4 192.168.201.11 t=0 0 m=audio 52322 RTP/AVP 0 c=IN IP4 [provider public IP] a=rtpmap:0 PCMU/8000 The "200 OK" arrives with similar SDP. Note that there are two connection addresses in the SDP, one private (the provider's -- I'm not using NAT) and one public. The problem is that Asterisk attempts to send media to the private address; of course, that doesn't work. If I use the provider-supplied ATA, or a Cisco ATA, it works fine. The SDP is similar, but the ATAs know to send media to the correct public IP. At first, I thought that the incoming SDP was improper, but RFC 2327 says: A session announcement must contain one "c=" field in each media description (see below) or a "c=" field at the session-level. It may contain a session-level "c=" field and one additional "c=" field per media description, in which case the per-media values override the session-level settings for the relevant media. So, it appears that Asterisk is not interpreting the SDP correctly. I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug has already been fixed in a later version (I can't find anything that seems relevant at bugs.digium.com)? If so, is there an easy way to upgrade [EMAIL PROTECTED] from the CVS? If not, could someone please suggest where to start looking at the code? Thanks, Stewart _______________________________________________ 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
