I had the same problem when we were setting up these boxes after
katrina. What i found is that they will only do one G729 session at a
time. so that mesg that your showing is that its trying to
register two chans as 729. what i did to get around this was to
turn off fource prefered codec on one line. This threw me for a loop
also but trust me this is the fix, and yes you can only make one 729
call at a time.<br>
<br>
<br>
Jason Price<br><br><div><span class="gmail_quote">On 11/23/05, <b class="gmail_sendername">Aaron Clauson</b> <<a href="mailto:aza@azaclauson.com">aza@azaclauson.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br><br>I have a very strange Asterisk SIP call signalling problem that is proving<br>extremely difficult to track down. The problem is that any SIP INVITE<br>request that is coming into Asterisk over a satellite connection from a
<br>Linksys Router or PAP2 is getting a "Not Acceptable Here (codec error)" from<br>Asterisk. I've done all the normal checks on the allowed codecs in sip.conf<br>but to no avail.<br><br>I've even gone as far as writing a basic SIP stack to authenticate and send
<br>the INVITE request to Asterisk with exactly the same SDP payload to let me<br>brute force different options in the SDP request to try an narrow it down<br>that way. The preplexing thing from that length exercise is that if exactly
<br>the same INVITE request comes in from my app across the same satellite<br>connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2 or<br>WRT54GP2 it gets 488 Codec Not Acceptable Here'ed.<br><br>The first time this happened we went through all the usual checks and got
<br>nowhere and the person drifted off and it was put down to something speicifc<br>to that set up/connection. But now it's cropped up again with a different<br>person who also just happens to be on a satellite connection but from a
<br>different provider, although it is possible both providers use the same<br>infrastructure. In both cases incoming calls to the Linksys devices worked<br>correctly it's just the outgoing calls from the devices to Asterisk that are
<br>getting the rejection. In the second case we can't put it down to something<br>to do with the connection because the person has a Vonage service working no<br>problems across the same satellite link we are getting the rejection on.
<br><br>The SIP trace is below and I'm wondering if anybody has ever seen something<br>similar. The only thing I can think of is that it's somehow a timing issue I<br>can't see how it can be a codec issue since the exactly the same SDP payload
<br>will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive to<br>the any timings in the INVITE request? It seems highly unlikely but I just<br>can't think of anything else.<br><br>INVITE <a href="mailto:sip:018XXX@sip.XXX">
sip:018XXX@sip.XXX</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.248:5061">192.168.1.248:5061</a>;branch=z9hG4bK-3b91173f<br>From: XXX <<a href="mailto:sip:XXX@sip.XXX">sip:XXX@sip.XXX</a>>;tag=831f2cca367c3ddfo1
<br>To: <<a href="mailto:sip:018XXX@sip.xxx">sip:018XXX@sip.xxx</a>><br>Call-ID: <a href="mailto:c71dab66-43f06ff3@192.168.1.248">c71dab66-43f06ff3@192.168.1.248</a><br>CSeq: 103 INVITE<br>Max-Forwards: 70<br>Proxy-Authorization: Digest
<br>username="XXX",realm="asterisk",nonce="489bfe04",uri="<a href="mailto:sip:018XXX@sip.XXX">sip:018XXX@sip.XXX</a>",al<br>gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
<br>Contact: XXX <sip:XXX@192.168.1.248:5061><br>Expires: 240<br>User-Agent: Linksys/PAP2-3.1.3(LS)<br>Content-Length: 424<br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<br>Supported: x-sipura<br>Content-Type: application/sdp
<br><br>v=0<br>o=- 418210 418210 IN IP4 <a href="http://192.168.1.248">192.168.1.248</a><br>s=-<br>c=IN IP4 <a href="http://192.168.1.248">192.168.1.248</a><br>t=0 0<br>m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:2 G726-32/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729a/8000<br>a=rtpmap:96 G726-40/8000<br>a=rtpmap:97 G726-24/8000<br>a=rtpmap:98 G726-16/8000<br>a=rtpmap:100 NSE/8000
<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:30<br>a=sendrecv<br><br>----------------------------------------------------------------------------<br>----<br><br>SIP/2.0 407 Proxy Authentication Required
<br>Via: SIP/2.0/UDP<br><a href="http://192.168.1.248:5061">192.168.1.248:5061</a>;branch=z9hG4bK-3b91173f;received=xxx;rport=5061<br>From: xxx <<a href="mailto:sip:XXX@sip.xxx">sip:XXX@sip.xxx</a>>;tag=831f2cca367c3ddfo1
<br>To: <<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>>;tag=as17d663fb<br>Call-ID: <a href="mailto:c71dab66-43f06ff3@192.168.1.248">c71dab66-43f06ff3@192.168.1.248</a><br>CSeq: 103 INVITE<br>User-Agent: asterisk
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY<br>Contact: <sip:018xxx@xxx><br>Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3"<br>Content-Length: 0<br><br><br>----------------------------------------------------------------------------
<br>----<br><br>ACK <a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.248:5061">192.168.1.248:5061</a>;branch=z9hG4bK-c341696b<br>From: xxx <<a href="mailto:sip:xxx@sip.xxx">
sip:xxx@sip.xxx</a>>;tag=831f2cca367c3ddfo1<br>To: <<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>>;tag=as50c8f92d<br>Call-ID: <a href="mailto:c71dab66-43f06ff3@192.168.1.248">c71dab66-43f06ff3@192.168.1.248
</a><br>CSeq: 102 ACK<br>Max-Forwards: 70<br>Proxy-Authorization: Digest<br>username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>
",al<br>gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f"<br>Contact: xxx <sip:xxx@192.168.1.248:5061><br>User-Agent: Linksys/PAP2-3.1.3(LS)<br>Content-Length: 0<br><br><br>----------------------------------------------------------------------------
<br>----<br><br>INVITE <a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.248:5061">192.168.1.248:5061</a>;branch=z9hG4bK-3b91173f<br>From: xxx <<a href="mailto:sip:xxx@sip.xxx">
sip:xxx@sip.xxx</a>>;tag=831f2cca367c3ddfo1<br>To: <<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>><br>Call-ID: <a href="mailto:c71dab66-43f06ff3@192.168.1.248">c71dab66-43f06ff3@192.168.1.248</a><br>
CSeq: 103 INVITE<br>Max-Forwards: 70<br>Proxy-Authorization: Digest<br>username="xxx",realm="asterisk",nonce="489bfe04",uri="<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>",al
<br>gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"<br>Contact: xxx <sip:xxx@192.168.1.248:5061><br>Expires: 240<br>User-Agent: Linksys/PAP2-3.1.3(LS)<br>Content-Length: 424<br>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
<br>Supported: x-sipura<br>Content-Type: application/sdp<br><br>v=0<br>o=- 418210 418210 IN IP4 <a href="http://192.168.1.248">192.168.1.248</a><br>s=-<br>c=IN IP4 <a href="http://192.168.1.248">192.168.1.248</a><br>t=0 0
<br>m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:2 G726-32/8000<br>a=rtpmap:4 G723/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729a/8000<br>a=rtpmap:96 G726-40/8000<br>a=rtpmap:97 G726-24/8000
<br>a=rtpmap:98 G726-16/8000<br>a=rtpmap:100 NSE/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:30<br>a=sendrecv<br><br>----------------------------------------------------------------------------
<br>----<br><br>SIP/2.0 488 Not Acceptable Here (codec error)<br>Via: SIP/2.0/UDP<br><a href="http://192.168.1.248:5061">192.168.1.248:5061</a>;branch=z9hG4bK-3b91173f;received=xxx;rport=5061<br>From: xxx <<a href="mailto:sip:xxx@sip.xxx">
sip:xxx@sip.xxx</a>>;tag=831f2cca367c3ddfo1<br>To: <<a href="mailto:sip:018xxx@sip.xxx">sip:018xxx@sip.xxx</a>>;tag=as17d663fb<br>Call-ID: <a href="mailto:c71dab66-43f06ff3@192.168.1.248">c71dab66-43f06ff3@192.168.1.248
</a><br>CSeq: 103 INVITE<br>User-Agent: asterisk<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY<br>Contact: <sip:018xxx@xxx><br>Content-Length: 0<br><br>Thanks,<br><br>Aaron<br><br><br>_______________________________________________
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