[Asterisk-Users] PAP2 won't make two g729 calls at the same time

[Warren Burstein]

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:59342 at 192.168.121.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3e521aa
From: PAP 220 <sip:220 at 192.168.121.20>;tag=6b66e68deef168b2o0
To: <sip:59342 at 192.168.121.20>
Call-ID: 8e8903e9-18188b06 at 192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220 at 192.168.254.44:5060>
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:203 at 192.168.121.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15
From: PAP 220 <sip:220 at 192.168.121.20>;tag=b8b86be991749af5o0
To: <sip:203 at 192.168.121.20>
Call-ID: a44265f9-c09c6825 at 192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220 at 192.168.254.44:5060>
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