thr3ads.net - asterisk users - [Asterisk-Users] PAP2 won't make two g729 calls at the same time [Mar 2006]

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Warren Burstein

2006-Mar-08 10:43 UTC

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

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@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@192.168.121.20>;tag=6b66e68deef168b2o0
To: <sip:59342@192.168.121.20>
Call-ID: 8e8903e9-18188b06@192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220@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@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@192.168.121.20>;tag=b8b86be991749af5o0
To: <sip:203@192.168.121.20>
Call-ID: a44265f9-c09c6825@192.168.254.44
CSeq: 101 INVITE
Max-Forwards: 70
Contact: PAP 220 <sip:220@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

Joseph Tanner

2006-Mar-08 18:37 UTC

head link

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

The PAP2 can only handle one g729 call at one time.  Whether that's a
hardware limitation, or licensing, or both, I don't know.

Joseph Tanner

On 3/8/06, Warren Burstein <warren@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@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@192.168.121.20>;tag=6b66e68deef168b2o0
> To: <sip:59342@192.168.121.20>
> Call-ID: 8e8903e9-18188b06@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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@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@192.168.121.20>;tag=b8b86be991749af5o0
> To: <sip:203@192.168.121.20>
> Call-ID: a44265f9-c09c6825@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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
>
>
>
>
> _______________________________________________
> --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

2006-Mar-08 18:42 UTC

head link

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

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@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@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@192.168.121.20>;tag=6b66e68deef168b2o0
> To: <sip:59342@192.168.121.20>
> Call-ID: 8e8903e9-18188b06@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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@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@192.168.121.20>;tag=b8b86be991749af5o0
> To: <sip:203@192.168.121.20>
> Call-ID: a44265f9-c09c6825@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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
>
>
>
>
> _______________________________________________
> --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

Nick Hoffman

2006-Mar-08 18:56 UTC

head link

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

On Thu March 9 2006 03:43, Warren Burstein <warren@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?

Hi Warren. On the PAP2, if you can make 2 simultaneous calls but only 1 can 
use G.729, I would hazard a guess that the PAP2 only has 1 G.729 license 
installed on it. I doubt that can be increased.

Hope that helps.
-- Nick
e: nick.hoffman@altcall.com
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
use of the email.  We do not waive any privilege, confidentiality or 
copyright associated with it.

Leo Ann Boon

2006-Mar-08 21:37 UTC

head link

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

Warren Burstein 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?
It's a PAP2 feature. The PAP2 hardware is only capable of 1 (ONE) G.729 
call at any time. The limit also applies if you're doing conferencing on 
the PAP2.
>
> 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@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@192.168.121.20>;tag=6b66e68deef168b2o0
> To: <sip:59342@192.168.121.20>
> Call-ID: 8e8903e9-18188b06@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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@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@192.168.121.20>;tag=b8b86be991749af5o0
> To: <sip:203@192.168.121.20>
> Call-ID: a44265f9-c09c6825@192.168.254.44
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: PAP 220 <sip:220@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
>
>
>
>
> _______________________________________________
> --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
>

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