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

[Tom Vile]

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 <warren at softov.co.il> 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: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
>
>
>
>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax:     518-631-2856