This ATA can only do 1 g729 call at a time. The sipura 2002 is the same way. It's outlined in the datasheet.
On 3/8/06, Warren Burstein <[EMAIL PROTECTED]> wrote: > I have a Linksys PAP2. Identical setups for the two channels in both > the unit and in Asterisk. In particular, both channels enable g729 and > set it as the preferred codec, and have disallow=all and allow=g729 in > sip.conf. > > If we make a call on one channel, it works (and uses g729), but if we > make a call on the other channel when the first one is still connected, > it fails. We have three g729 licenses, and no others were in use at the > times this happened, but even if we didn't have enough, how would the > PAP2 know that? > > Here's a good, and a bad INVITE message, from the log file with sip > debug enabled. Has anyone seen anything like this? > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa > From: PAP 220 <sip:[EMAIL PROTECTED]>;tag=6b66e68deef168b2o0 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 101 INVITE > Max-Forwards: 70 > Contact: PAP 220 <sip:[EMAIL PROTECTED]:5060> > Expires: 240 > User-Agent: Linksys/PAP2-3.1.3(LS) > Content-Length: 246 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > > v=0 > o=- 261305180 261305180 IN IP4 192.168.254.44 > s=- > c=IN IP4 192.168.254.44 > t=0 0 > m=audio 16392 RTP/AVP 18 100 101 > a=rtpmap:18 G729a/8000 > a=rtpmap:100 NSE/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15 > From: PAP 220 <sip:[EMAIL PROTECTED]>;tag=b8b86be991749af5o0 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 101 INVITE > Max-Forwards: 70 > Contact: PAP 220 <sip:[EMAIL PROTECTED]:5060> > Expires: 240 > User-Agent: Linksys/PAP2-3.1.3(LS) > Content-Length: 267 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > Content-Type: application/sdp > > v=0 > o=- 261589835 261589835 IN IP4 192.168.254.44 > s=- > c=IN IP4 192.168.254.44 > t=0 0 > m=audio 16400 RTP/AVP 0 8 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:100 NSE/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users