RE: [Serusers] Voicemail wierdness

24 Aug 2004

Update....

I figured out why it was screwing up with the offline users... that
append_branch(); statement was screwing it up (sensible when u think
about it really).  Take it out all is fine.

However, still struggling with correct technique for using mediaproxy
with vm.   It seems to be mainly be a problem with failure routes.....
If anyone has any immediate suggestions, let me know.... Otherwise I
will have a more detailed look tomorrow and post a more detailed
description.

On a further note, I have an interesting conundrum... hopefully someone
has found a way round this:

All users have an alphanumeric address + a 4 digit numerical alias.  I
am trying to use asterisk for vm primarily because it allows users to
dial in and retrieve their mail.  To that end, I decided to store their
VM boxes as the numerical alias.  Herein lies the problem... If I call,
for example, 1000@xxxxx then vm works correctly - the 1000 is passed to
asterisk which does the lookup.  If I call admin@xxxxxxx then obviously
it doesn't work.... The vm box admin does not exist in asterisk.   If I
miss off the revert_uri(); statement, things are slightly better, but
the wrong way around... if I call 1000, the alias lookup takes place and
admin@xxxxx is passed to asterisk.   However, this does not help me...
as although always passing asterisk the same username, if a user dials
in they cant enter "admin" as the mailbox from a normal phone. What I
really need is the opposite of revert_uri in this case... something
which will always use the alias...I could use some  external script to
do a dblookup and then modify the uri but that seems a bit messy, im
sure there must be a way in ser to do it... any ideas are very welcome.

Many tnanks again,

Dave

________________________________

From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Dave Bath
Sent: 24 August 2004 17:02
To: serusers(a)lists.iptel.org
Subject: [Serusers] Voicemail wierdness

Hey all,

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:

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;
}

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?).

However, i have two problems:

(1) If the call is from Nonnat  UA --> SER --> Mediaproxy --> Nat UA

                                                      |

                                                     \/

                                               Asterisk VM

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?

(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....

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!

Dave

U 161.30.94.151:2378 -> 161.30.94.136:5060
  INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..T
  o: sip:1000@sip.dev.inmarsat.com..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 1 INVITE..Cont
  act: <sip:test3@161.30.94.151>..Content-Type:
application/sdp..Content-Length: 308..Accept-Langua
  ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
REGISTER, SUBSCRIBE..Supported:
  sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11
(VxWorks)..Date: Tue, 24 Aug 2004
  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-
  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
  0/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0 pcmu/8000/1..a=rtpmap:8
pcma/8000/1..a=rtpmap:18 g729/
  8000/1..a=fmtp:18 annexb=no..a=rtpmap:98 telephone-event/8000/1..
#
U 161.30.94.136:5060 -> 161.30.94.151:5060
  SIP/2.0 407 Proxy Authentication Required..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:

sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c718b0.21f2.
.Call-Id: call-1093359137
  -28@161.30.94.151..Cseq: 1 INVITE..Via: SIP/2.0/UDP
161.30.94.151..Proxy-Authenticate: Digest rea
  lm="sip.dev.inmarsat.com",
nonce="412b574f5e5a864751e3e96fb32f3f7a4929a17a"..Server: Sip EXpress
  router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
161.30.94.136:5060 "Noisy feedback
  tells:  pid=25819 req_src_ip=161.30.94.151 req_src_port=2378
in_uri=sip:1000@sip.dev.inmarsat.com
   out_uri=sip:admin@sip.dev.inmarsat.com via_cnt==1"....
#
U 161.30.94.151:2378 -> 161.30.94.136:5060
  ACK sip:1000@sip.dev.inmarsat.com SIP/2.0..Contact:
sip:test3@161.30.94.151..From: sip:test3@sip.
  dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=9d060f1cca80def62c99c3e616c71
  8b0.21f2..Call-Id: call-1093359137-28@161.30.94.151..Cseq: 1
ACK..Accept-Language: en..User-Agent
  : Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via:
SIP/2.0/UDP 161.30.94.151..
  Content-Length: 0....
#
U 161.30.94.151:2378 -> 161.30.94.136:5060
  INVITE sip:1000@sip.dev.inmarsat.com SIP/2.0..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..T
  o: sip:1000@sip.dev.inmarsat.com..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Cont
  act: <sip:test3@161.30.94.151>..Content-Type:
application/sdp..Content-Length: 308..Accept-Langua
  ge: en..Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY,
REGISTER, SUBSCRIBE..Supported:
  sip-cc, sip-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11
(VxWorks)..Date: Tue, 24 Aug 2004
  14:52:19 GMT..Proxy-Authorization: DIGEST USERNAME="test3",
REALM="sip.dev.inmarsat.com", NONCE="
  412b574f5e5a864751e3e96fb32f3f7a4929a17a",
URI="sip:1000@sip.dev.inmarsat.com", RESPONSE="520ee04
  3cedf1c3d2ccbc0fa05101d2c"..Via: SIP/2.0/UDP
161.30.94.151....v=0..o=Pingtel 5 5 IN IP4 161.30.94
  .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
  :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0
pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r
  tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98
telephone-event/8000/1..
#
U 161.30.94.136:5060 -> 161.30.94.151:5060
  SIP/2.0 100 trying -- your call is important to us..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24
  948..To: sip:1000@sip.dev.inmarsat.com..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 2 INVITE
  ..Via: SIP/2.0/UDP 161.30.94.151..Server: Sip EXpress router (0.8.14
(i386/linux))..Content-Lengt
  h: 0..Warning: 392 161.30.94.136:5060 "Noisy feedback tells:
pid=25827 req_src_ip=161.30.94.151
  req_src_port=2378 in_uri=sip:1000@sip.dev.inmarsat.com
out_uri=sip:1000@asterisk.dev.inmarsat.com
  :5060 via_cnt==1"....
#
U 161.30.94.136:5060 -> 161.30.94.137:5060
  INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards:
10..Record-Route: <sip:1000
  @161.30.94.136;ftag=2c24948;lr=on>..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100
  0@sip.dev.inmarsat.com..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip:
  test3@161.30.94.151>..Content-Type
<mailto:test3@161.30.94.151%3e..Content-Type> :
application/sdp..Content-Length: 308..Accept-Language: en..Al
  low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported: sip-cc, si
  p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date:
Tue, 24 Aug 2004 14:52:19 G
  MT..Proxy-Authorization: DIGEST USERNAME="test3",
REALM="sip.dev.inmarsat.com", NONCE="412b574f5e
  5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com",
RESPONSE="520ee043cedf1c3d2
  ccbc0fa05101d2c"..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP
  161.30.94.151..CC-Diversion:
sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94
  .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
  :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0
pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r
  tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98
telephone-event/8000/1..
#
U 161.30.94.136:5060 -> 161.30.94.137:5060
  INVITE sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Max-Forwards:
10..Record-Route: <sip:1000
  @161.30.94.136;ftag=2c24948;lr=on>..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:100
  0@sip.dev.inmarsat.com..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 2 INVITE..Contact: <sip:
  test3@161.30.94.151>..Content-Type
<mailto:test3@161.30.94.151%3e..Content-Type> :
application/sdp..Content-Length: 308..Accept-Language: en..Al
  low: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE..Supported: sip-cc, si
  p-cc-01, timer, replaces..User-Agent: Pingtel/2.1.11 (VxWorks)..Date:
Tue, 24 Aug 2004 14:52:19 G
  MT..Proxy-Authorization: DIGEST USERNAME="test3",
REALM="sip.dev.inmarsat.com", NONCE="412b574f5e
  5a864751e3e96fb32f3f7a4929a17a", URI="sip:1000@sip.dev.inmarsat.com",
RESPONSE="520ee043cedf1c3d2
  ccbc0fa05101d2c"..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UDP
  161.30.94.151..CC-Diversion:
sip:1000@sip.dev.inmarsat.com....v=0..o=Pingtel 5 5 IN IP4 161.30.94
  .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
  :96 eg711u/8000/1..a=rtpmap:97 eg711a/8000/1..a=rtpmap:0
pcmu/8000/1..a=rtpmap:8 pcma/8000/1..a=r
  tpmap:18 g729/8000/1..a=fmtp:18 annexb=no..a=rtpmap:98
telephone-event/8000/1..
#
U 161.30.94.137:5060 -> 161.30.94.136:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UD
  P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.
  com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2
INVITE..User-Agent: Asteri
  sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:1000@161.30.94.137>..Conte
  nt-Length: 0....
#
U 161.30.94.137:5060 -> 161.30.94.136:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.1..Via: SIP/2.0/UD
  P 161.30.94.151..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.
  com;tag=as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2
INVITE..User-Agent: Asteri
  sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:1000@161.30.94.137>..Conte
  nt-Length: 0....
#
U 161.30.94.137:5060 -> 161.30.94.136:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.0..Via: SIP/2.0/UDP 16
  1.30.94.151..Record-Route:
<sip:1000@161.30.94.136;ftag=2c24948;lr=on>..From: sip:test3@sip.dev.i
  nmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10933591
  37-28@161.30.94.151..CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow:
INVITE, ACK, CANCEL, OPTION
  S, BYE, REFER..Contact: <sip:1000@161.30.94.137>..Content-Type:
application/sdp..Content-Length:
  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
  dio 10248 RTP/AVP 18 0 97 3 98..a=rtpmap:18 G729/8000..a=rtpmap:0
PCMU/8000..a=rtpmap:97 iLBC/800
  0..a=rtpmap:3 GSM/8000..a=rtpmap:98 telephone-event/8000..a=fmtp:98
0-16..a=silenceSupp:off - - -
   -..
#
U 161.30.94.136:5060 -> 161.30.94.151:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 161.30.94.151..Record-Route:
<sip:1000@161.30.94.136;ftag=2c2494
  8;lr=on>..From: sip:test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag
  =as78a37589..Call-ID: call-1093359137-28@161.30.94.151..CSeq: 2
INVITE..User-Agent: Asterisk PBX.
  .Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Contact:
<sip:1000@161.30.94.137>..Content-Type
  : application/sdp..Content-Length: 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=audio 10248 RTP/AVP 18 0 97 3
98..a=rtpmap:18 G729/8000..a=rtpma
  p:0 PCMU/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:3 GSM/8000..a=rtpmap:98
telephone-event/8000..a=fm
  tp:98 0-16..a=silenceSupp:off - - - -..
#
U 161.30.94.136:5060 -> 161.30.94.137:5060
  CANCEL sip:1000@asterisk.dev.inmarsat.com:5060 SIP/2.0..Via:
SIP/2.0/UDP 161.30.94.136;branch=z9h
  G4bK4773.683f9521.1..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..Call-Id: call-1093359137-2
  8@161.30.94.151..To: sip:1000@sip.dev.inmarsat.com..Cseq: 2
CANCEL..User-Agent: Sip EXpress route
  r(0.8.14 (i386/linux))..Content-Length: 0....
#
U 161.30.94.137:5060 -> 161.30.94.136:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
161.30.94.136;branch=z9hG4bK4773.683f9521.1..From: sip:test3@sip
  .dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-ID: call-10
  93359137-28@161.30.94.151..CSeq: 2 CANCEL..User-Agent: Asterisk
PBX..Allow: INVITE, ACK, CANCEL,
  OPTIONS, BYE, REFER..Contact:
<sip:1000@161.30.94.137>..Content-Length: 0....
#
U 161.30.94.151:2394 -> 161.30.94.136:5060
  ACK sip:1000@161.30.94.136;ftag=2c24948;lr=on SIP/2.0..Route:
<sip:1000@161.30.94.137>..Contact:
  sip:test3@161.30.94.151..From:
sip:test3@sip.dev.inmarsat.com;tag=2c24948..To: sip:1000@sip.dev.i
  nmarsat.com;tag=as78a37589..Call-Id:
call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Langua
  ge: en..User-Agent: Pingtel/2.1.11 (VxWorks)..Date: Tue, 24 Aug 2004
14:52:19 GMT..Via: SIP/2.0/U
  DP 161.30.94.151..Content-Length: 0....
#
U 161.30.94.136:5060 -> 161.30.94.137:5060
  ACK sip:1000@161.30.94.137 SIP/2.0..Max-Forwards: 10..Contact:
sip:test3@161.30.94.151..From: sip
  :test3@sip.dev.inmarsat.com;tag=2c24948..To:
sip:1000@sip.dev.inmarsat.com;tag=as78a37589..Call-I
  d: call-1093359137-28@161.30.94.151..Cseq: 2 ACK..Accept-Language:
en..User-Agent: Pingtel/2.1.11
   (VxWorks)..Date: Tue, 24 Aug 2004 14:52:19 GMT..Via: SIP/2.0/UDP
161.30.94.136;branch=0..Via: SI
  P/2.0/UDP 161.30.94.151..Content-Length: 0....
exit

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RE: [Serusers] Voicemail wierdness