<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML DIR=ltr><HEAD><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"></HEAD><BODY><DIV><FONT face='Arial' color=#000000 size=2>Hey all,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Another wierdness for a sunny tuesday
afternoon.... Have been using ser and * quite successfully so far, and am
not into the stage of testing voicemail. Currently I am using *
with mysql vm storage. I have a failure route which affter the
fr_timer expires, does:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><!--StartFragment --> <PRE>failure_route[4] {
# If we've got here, it means that a call timer has expired, and the user is in the
# voicemail group. Therefore we're going to try and hand off the call to the Asterisk
# voicemail system.
revert_uri();
rewritehostport("asterisk.dev.inmarsat.com:5060");
log (1, "VOICEMAIL: VM-Divert --> Handing off to Asterisk\n");
append_branch();
append_urihf("CC-Diversion: ", "\r\n");
t_relay();
break;
}</PRE></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>This works great for normal users who are
registered but not answering their phone (or indeed are busy it seems - i
havent' worked out how to configure this behaviour -- any
suggestions?).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>However, i have two problems:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>(1) If the call is from Nonnat UA --> SER
--> Mediaproxy --> Nat UA</FONT></DIV>
<DIV><FONT face=Arial
size=2>
|</FONT></DIV>
<DIV><FONT face=Arial
size=2> \/</FONT></DIV>
<DIV><FONT face=Arial
size=2>
Asterisk VM</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>then the Nonnat UA doesn't get audio from asterisk
vm. I think this has somethign to do with mediaproxy trying to proxy the vm
request when it shouldnt and not knwoing where to send it - again, any
suggestions?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>(2) The most annoying issue at the moment is for
offline users. It seems logical that if you have an offline user (one that
has been found in the subscriber db but just isnt' registered at the moment) you
might want their address to divert to vm. However, no amount of fiddling
has enabled be to use a failure route for this, and i'm not sure why. In
desperation i just created a new route block to handle it - the same as above,
but called route[4] instead of failure_route. This at least forwards the
request correctly. However, all i get is "The p...." and then the media stream
cuts out, and * console shows the channel has been hung up. Below is the
ngrep. For some reason two INVITE reuqests are sent from ser(.136)
to *(.137) and i'm not sure why. Notice also that a CANCEL is generated,
which corresponds to the media stream dying almost as soon as it starts, but
again i'm not sure why....</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>If anyone has anyideas... if you need to look at
the ser.cfg then let me know. I'm sure there must be a mor elegant way of
handling vm but im not sure what it is at the moment!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Dave</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>U 161.30.94.151:2378 -> 161.30.94.136:5060<BR> INVITE
sip:1000@sip.dev.inmarsat.com SIP/2.0..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..T<BR> o:
sip:1000@sip.dev.inmarsat.com..Call-Id: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
1 INVITE..Cont<BR> act: <sip:test3@161.30.94.151>..Content-Type:
application/sdp..Content-Length: 308..Accept-Langua<BR> ge: en..Allow:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported:<BR> sip-cc, sip-cc-01, timer, replaces..User-Agent:
Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004<BR> 14:52:19 GMT..Via:
SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5 IN IP4
161.30.94.151..s=phone-<BR> call..c=IN IP4 161.30.94.151..t=0 0..m=audio
8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap:96 eg711u/800<BR> 0/1..a=rtpmap:97
eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=rtpmap:18
g729/<BR> 8000/1..a=fmtp:18 annexb=no..a=rtpmap:98
telephone-event/8000/1..<BR>#<BR>U 161.30.94.136:5060 ->
161.30.94.151:5060<BR> SIP/2.0 407 Proxy Authentication Required..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:<BR>
sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c718b0.21f2..Call-Id:
call-1093359137<BR> <A
href="mailto:-28@161.30.94.151..Cseq">-28@161.30.94.151..Cseq</A>: 1
INVITE..Via: SIP/2.0/UDP 161.30.94.151..Proxy-Authenticate: Digest rea<BR>
lm="sip.dev.inmarsat.com",
nonce="412b574f5e5a864751e3e96fb32f3f7a4929a17a"..Server: Sip EXpress<BR>
router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 161.30.94.136:5060
"Noisy feedback<BR> tells: pid=25819 req_src_ip=161.30.94.151
req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com<BR>
out_uri=sip:admin@sip.dev.inmarsat.com via_cnt==1"....<BR>#<BR>U
161.30.94.151:2378 -> 161.30.94.136:5060<BR> ACK
sip:1000@sip.dev.inmarsat.com SIP/2.0..Contact: sip:test3@161.30.94.151..From:
sip:test3@sip.<BR> dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c71<BR>
8b0.21f2..Call-Id: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
1 ACK..Accept-Language: en..User-Agent<BR> : Pingtel/2.1.11
(VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP
161.30.94.151..<BR> Content-Length: 0....<BR>#<BR>U 161.30.94.151:2378
-> 161.30.94.136:5060<BR> INVITE sip:1000@sip.dev.inmarsat.com
SIP/2.0..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..T<BR> o:
sip:1000@sip.dev.inmarsat.com..Call-Id: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 INVITE..Cont<BR> act: <sip:test3@161.30.94.151>..Content-Type:
application/sdp..Content-Length: 308..Accept-Langua<BR> ge: en..Allow:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported:<BR> sip-cc, sip-cc-01, timer, replaces..User-Agent:
Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004<BR> 14:52:19
GMT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com",
NONCE="<BR> 412b574f5e5a864751e3e96fb32f3f7a4929a17a",
URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee04<BR>
3cedf1c3d2ccbc0fa05101d2c"..Via: SIP/2.0/UDP 161.30.94.151....v=0..o=Pingtel 5 5
IN IP4 161.30.94<BR> .151..s=phone-call..c=IN IP4 161.30.94.151..t=0
0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap<BR> :96
eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8
pcma/8000/1..a=r<BR> tpmap:18 g729/8000/1..a=fmtp:18
annexb=no..a=rtpmap:98 telephone-event/8000/1..<BR>#<BR>U 161.30.94.136:5060
-> 161.30.94.151:5060<BR> SIP/2.0 100 trying -- your call is important
to us..From: sip:test3@sip.dev.inmarsat.com;tag=2c24<BR> 948..To:
sip:1000@sip.dev.inmarsat.com..Call-Id: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 INVITE<BR> ..Via: SIP/2.0/UDP 161.30.94.151..Server: Sip EXpress router
(0.8.14 (i386/linux))..Content-Lengt<BR> h: 0..Warning: 392
161.30.94.136:5060 "Noisy feedback tells: pid=25827
req_src_ip=161.30.94.151<BR> req_src_port=2378
in_uri=sip:1000@sip.dev.inmarsat.com
out_uri=sip:1000@asterisk.dev.inmarsat.com<BR> :5060
via_cnt==1"....<BR>#<BR>U 161.30.94.136:5060 -> 161.30.94.137:5060<BR>
INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards:
10..Record-Route: <sip:1000<BR>
@161.30.94.136;ftag=2c24948;lr=on>..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100<BR> <A
href="mailto:0@sip.dev.inmarsat.com..Call-Id">0@sip.dev.inmarsat.com..Call-Id</A>:
<A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 INVITE..Contact: <sip:<BR> <A
href="mailto:test3@161.30.94.151>..Content-Type">test3@161.30.94.151>..Content-Type</A>:
application/sdp..Content-Length: 308..Accept-Language: en..Al<BR> low:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported: sip-cc, si<BR> p-cc-01, timer, replaces..User-Agent:
Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G<BR>
MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com",
NONCE="412b574f5e<BR> 5a864751e3e96fb32f3f7a4929a17a",
URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2<BR>
ccbc0fa05101d2c"..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP<BR>
161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5
IN IP4 161.30.94<BR> .151..s=phone-call..c=IN IP4 161.30.94.151..t=0
0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap<BR> :96
eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8
pcma/8000/1..a=r<BR> tpmap:18 g729/8000/1..a=fmtp:18
annexb=no..a=rtpmap:98 telephone-event/8000/1..<BR>#<BR>U 161.30.94.136:5060
-> 161.30.94.137:5060<BR> INVITE
sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards: 10..Record-Route:
<sip:1000<BR> @161.30.94.136;ftag=2c24948;lr=on>..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100<BR> <A
href="mailto:0@sip.dev.inmarsat.com..Call-Id">0@sip.dev.inmarsat.com..Call-Id</A>:
<A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 INVITE..Contact: <sip:<BR> <A
href="mailto:test3@161.30.94.151>..Content-Type">test3@161.30.94.151>..Content-Type</A>:
application/sdp..Content-Length: 308..Accept-Language: en..Al<BR> low:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported: sip-cc, si<BR> p-cc-01, timer, replaces..User-Agent:
Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 G<BR>
MT..Proxy-Authorization: DIGEST USERNAME="test3", REALM="sip.dev.inmarsat.com",
NONCE="412b574f5e<BR> 5a864751e3e96fb32f3f7a4929a17a",
URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee043cedf1c3d2<BR>
ccbc0fa05101d2c"..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UDP<BR>
161.30.94.151..CC-Diversion: sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5
IN IP4 161.30.94<BR> .151..s=phone-call..c=IN IP4 161.30.94.151..t=0
0..m=audio 8766 RTP/AVP 96 97 0 8 18 98..a=rtpmap<BR> :96
eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8
pcma/8000/1..a=r<BR> tpmap:18 g729/8000/1..a=fmtp:18
annexb=no..a=rtpmap:98 telephone-event/8000/1..<BR>#<BR>U 161.30.94.137:5060
-> 161.30.94.136:5060<BR> SIP/2.0 100 Trying..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UD<BR> P
161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.<BR> com;tag=as78a37589..Call-ID: <A
href="mailto:call-1093359137-28@161.30.94.151..CSeq">call-1093359137-28@161.30.94.151..CSeq</A>:
2 INVITE..User-Agent: Asteri<BR> sk PBX..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte<BR>
nt-Length: 0....<BR>#<BR>U 161.30.94.137:5060 -> 161.30.94.136:5060<BR>
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UD<BR> P
161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.<BR> com;tag=as78a37589..Call-ID: <A
href="mailto:call-1093359137-28@161.30.94.151..CSeq">call-1093359137-28@161.30.94.151..CSeq</A>:
2 INVITE..User-Agent: Asteri<BR> sk PBX..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Conte<BR>
nt-Length: 0....<BR>#<BR>U 161.30.94.137:5060 -> 161.30.94.136:5060<BR>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 16<BR>
1.30.94.151..Record-Route:
<sip:1000@161.30.94.136;ftag=2c24948;lr=on>..From:
sip:test3@sip.dev.i<BR> nmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10933591<BR>
<A href="mailto:37-28@161.30.94.151..CSeq">37-28@161.30.94.151..CSeq</A>: 2
INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTION<BR>
S, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type:
application/sdp..Content-Length:<BR> 288....v=0..o=root 2063 2063 IN IP4
161.30.94.137..s=session..c=IN IP4 161.30.94.137..t=0 0..m=au<BR> dio
10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpmap:0
PCMU/8000..a=rtpmap:97 iLBC/800<BR> 0..a=rtpmap:3 GSM/8000..a=rtpmap:98
telephone-event/8000..a=fmtp:98 0-16..a=silenceSupp:off - - -<BR>
-..<BR>#<BR>U 161.30.94.136:5060 -> 161.30.94.151:5060<BR> SIP/2.0 200
OK..Via: SIP/2.0/UDP 161.30.94.151..Record-Route:
<sip:1000@161.30.94.136;ftag=2c2494<BR> 8;lr=on>..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag<BR> =as78a37589..Call-ID: <A
href="mailto:call-1093359137-28@161.30.94.151..CSeq">call-1093359137-28@161.30.94.151..CSeq</A>:
2 INVITE..User-Agent: Asterisk PBX.<BR> .Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER..Contact:
<sip:1000@161.30.94.137>..Content-Type<BR> :
application/sdp..Content-Length: 288....v=0..o=root 2063 2063 IN IP4
161.30.94.137..s=session..<BR> c=IN IP4 161.30.94.137..t=0 0..m=audio
10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpma<BR> p:0
PCMU/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:3 GSM/8000..a=rtpmap:98
telephone-event/8000..a=fm<BR> tp:98 0-16..a=silenceSupp:off - - -
-..<BR>#<BR>U 161.30.94.136:5060 -> 161.30.94.137:5060<BR> CANCEL
sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Via: SIP/2.0/UDP
161.30.94.136;branch=z9h<BR> G4bK4773.683f9521.1..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..Call-Id: call-1093359137-2<BR>
<A href="mailto:8@161.30.94.151..To">8@161.30.94.151..To</A>:
sip:1000@sip.dev.inmarsat.com..Cseq: 2 CANCEL..User-Agent: Sip EXpress
route<BR> r(0.8.14 (i386/linux))..Content-Length: 0....<BR>#<BR>U
161.30.94.137:5060 -> 161.30.94.136:5060<BR> SIP/2.0 200 OK..Via:
SIP/2.0/UDP 161.30.94.136;branch=z9hG4bK4773.683f9521.1..From:
sip:test3@sip<BR> .dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10<BR> <A
href="mailto:93359137-28@161.30.94.151..CSeq">93359137-28@161.30.94.151..CSeq</A>:
2 CANCEL..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL,<BR>
OPTIONS, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Length:
0....<BR>#<BR>U 161.30.94.151:2394 -> 161.30.94.136:5060<BR> ACK
sip:1000@161.30.94.136;ftag=2c24948;lr=on SIP/2.0..Route:
<sip:1000@161.30.94.137>..Contact:<BR>
sip:test3@161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.i<BR> nmarsat.com;tag=as78a37589..Call-Id: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 ACK..Accept-Langua<BR> ge: en..User-Agent: Pingtel/2.1.11
(VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/U<BR> DP
161.30.94.151..Content-Length: 0....<BR>#<BR>U 161.30.94.136:5060 ->
161.30.94.137:5060<BR> ACK sip:1000@161.30.94.137 SIP/2.0..Max-Forwards:
10..Contact: sip:test3@161.30.94.151..From: sip<BR>
:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-I<BR> d: <A
href="mailto:call-1093359137-28@161.30.94.151..Cseq">call-1093359137-28@161.30.94.151..Cseq</A>:
2 ACK..Accept-Language: en..User-Agent: Pingtel/2.1.11<BR>
(VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP
161.30.94.136;branch=0..Via: SI<BR> P/2.0/UDP
161.30.94.151..Content-Length: 0....<BR>exit<BR></DIV>
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