Hello.
Asterisk "A" is version 1.2.1.
Asterisk "B" is version 1.0.9.
If I call by IAX from Asterisk "A" to B, and after that, Asterisk "B"
call by SIP to Openser, the call works.
The invite message from Asterisk to openser by Sip is:
U 2006/01/17 17:50:49.261265 10.2.11.50:5061 -> 10.2.11.50:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP
10.2.11.50:5061;branch=z9hG4bK722ced70..From: "Analogico" <sip:206@
10.2.11.50:5061>;tag=as4cdf4533..To: <sip:[EMAIL PROTECTED]>..Contact:
<sip:[EMAIL PROTECTED]:5061>..Call-ID: 2112bb831f0c32e
[EMAIL PROTECTED]: 102 INVITE..User-Agent: Asterisk
PBX..Date: Tue, 17 Jan 2006 16:50:49 GMT..Allow: I
NVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
application/sdp..Content-Length: 442....v=0..o=root 17529 17529
IN IP4 10.2.11.50..s=session..c=IN IP4 10.2.11.50..t=0 0..m=audio
12064 RTP/AVP 0 4 3 8 111 5 7 18 110 97 101..a=rtpmap
:0 PCMU/8000..a=rtpmap:4 G723/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:5 DV
I4/8000..a=rtpmap:7 LPC/8000..a=rtpmap:18 G729/8000..a=rtpmap:110
speex/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:101 telep
hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..
If I call by IAX from Asterisk "B" to Asterisk "A", and afterwords,
Asterisk "A" cal by SIP to openser, it fails.
The invite message from Asterisk to ser is:
U 2006/01/17 17:57:16.896269 10.2.11.35:5062 -> 10.2.11.35:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0..Via: SIP/2.0/UDP
10.2.11.35:5062;branch=z9hG4bK256111a9;rport..From: "David" <sip:"D
avid"<sip:[EMAIL PROTECTED]@10.2.11.35:5062>;tag=as2652cbb1..To:
<sip:[EMAIL PROTECTED]>..Contact: <sip:"David"<sip:[EMAIL PROTECTED]@
10.2.11.35:5062>..Call-ID:
[EMAIL PROTECTED]: 102
INVITE..User-Agent: Asterisk PBX..Max
-Forwards: 70..Date: Tue, 17 Jan 2006 16:57:16 GMT..Allow: INVITE,
ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
.Content-Type: application/sdp..Content-Length: 463....v=0..o=root
24590 24590 IN IP4 10.2.11.35..s=session..c=IN IP4 1
0.2.11.35..t=0 0..m=audio 16578 RTP/AVP 4 0 8 111 18 3 97 7 110 5
101..a=rtpmap:4 G723/8000..a=rtpmap:0 PCMU/8000..a=rt
pmap:8 PCMA/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 GSM/8000..a=rtpmap:
97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5
DVI4/8000..a=rtpmap:101 telephone-event/8000..a=
fmtp:101 0-16..a=silenceSupp:off - - - -..
This message is wrong. Which Asterisk works bad, 1.0.9 or 1.2.1?
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