[Kamailio-Users] Users Digest, Vol 41, Issue 24

luzango mfupe luzango.mfupe at gmail.com
Thu Oct 9 11:50:59 CEST 2008


Ok David, will give feedback soon.rgds,
Lu.

On Thu, Oct 9, 2008 at 11:53 AM, <users-request at lists.kamailio.org> wrote:

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> Today's Topics:
>
>   1. Re: No audio both ways. (David Villasmil)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 9 Oct 2008 11:07:12 +0200
> From: "David Villasmil" <david.villasmil.work at gmail.com>
> Subject: Re: [Kamailio-Users] No audio both ways.
> Cc: users at lists.kamailio.org
> Message-ID:
>        <9853f4ff0810090207p11d15bddt8dca8f57370c83c9 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Try adding a:
>
> nat=yes
>
> to the kamailio/openser peer definition and test
>
> dvg
>
> On Thu, Oct 9, 2008 at 10:58 AM, luzango mfupe <luzango.mfupe at gmail.com
> >wrote:
>
> >
> > Hi mates,I have this setup:
> > Xlite---->Openser---->Asterisk------>VoIP to PTSN Provider
> >
> > I use kamailio 1.3.3(port5060), MediaProxy and Asterisk 1.4 (port 5065)
> on
> > the same Debian Box in Realtime with no NAT. Asterisk connects calls to
> the
> > VoIP to PSTN provider. I am able to establish calls towards the PSTN
> > side(Landline & Mobiles)  but with no audio. I can hear the ringing tone
> but
> > when the call connects and the conversation begin i hear nothing so as
> the
> > Callee side.
> >
> > Below are my configs,the ngrep captured packets and codecs.
> >
> >
> ####################################################################################
> > route[4] {
> >     # routing to the public network
> >     rewritehostport("xx.xxx.xxx.xx:5065");
> >     t_on_failure("2");
> >     if (!t_relay()) {
> >         sl_reply_error();
> >     };
> > exit;
> > }
> >
> > route[6] {
> >     #
> >     # -- NAT handling --
> >     #
> >     if (isbflagset(6) || isbflagset(7)) {
> >         append_hf("P-hint: Route[6]: mediaproxy \r\n");
> >         use_media_proxy();
> >     };
> > }
> >
> > route[10] {
> >     #from an internal domain -> inbound
> >     #Native SIP destinations are handled using the location table
> >     #Gateway destinations are handled by regular expressions
> >     append_hf("P-hint: inbound->inbound \r\n");
> >
> >     if (uri=~"^sip:0[1-9][0-9]+ at .*") {
> >         if (is_user_in("credentials","local")) {
> >         strip(1);
> >         prefix("27");
> >            route(6);
> >            route(4);
> >             exit;
> >         } else {
> >             sl_send_reply("403", "No permissions for local calls");
> >             exit;
> >         };
> >     };
> >
> > if (uri=~"^sip:00[1-9][0-9]+ at .*") {
> >         if (is_user_in("credentials","int")) {
> >            strip(2);
> >            route(6);
> >             route(4);
> >             exit;
> >         } else {
> >             sl_send_reply("403", "No permissions for international
> calls");
> >         };
> >     };
> >
> >
> >
> ###################################################################################
> > This call was from the xlite softphone 1974 towards the Landline
> > 0123825710.
> >
> >
> ###################################################################################
> > U 2008/12/06 03:38:43.896057 196.212.209.18:46738 -> kamailio IP:5060
> >   INVITE sip:0123825710 at KAMAILIO ip SIP/2.0..Via: SIP/2.0/UDP
> 192.168.0.55
> > :
> >
> >
> 46738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..Max-Forward
> >   s: 70..Contact: <sip:1974 at 196.212.209.18:46738>..To:
> > "0123825710"<sip:01238
> >   25710 at kamailio IP>..From: <sip:1974 at kamailioIP>;tag=353dd217..Call-ID:
> >    MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 INVITE..Allow:
> > INVIT
> >   E, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
> > INFO..Cont
> >   ent-Type: application/sdp..User-Agent: X-Lite release 1014k stamp
> > 47051..Co
> >   ntent-Length: 237....v=0..o=- 1 2 IN IP4 192.168.0.55..s=CounterPath
> > X-Lite
> >    3.0..c=IN IP4 192.168.0.55..t=0 0..m=audio 60782 RTP/AVP 0 8 3
> > 101..a=fmtp
> >   :101 0-15..a=rtpmap:101 telephone-event/8000..a=alt:1 1 : ljiQYRpD
> > NiMZsfdZ
> >    192.168.0.55 60782..a=sendrecv..
> >
> > U 2008/12/06 03:38:43.896350 Kamailio IP:5060 -> 196.212.209.18:46738
> >   SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
> > 192.168.0.55:46
> >
> >
> 738;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport=46738;received
> >   =196.212.209.18..To: "0123825710"<sip:0123825710 at kamailioIP
> >;tag=329cfea
> >   a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974 at kamailioIP
> >;tag=353dd217
> >   ..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1
> > INVITE..Pr
> >   oxy-Authenticate: Digest realm="41.208.212.97",
> > nonce="4939d8cfeb060ab14354
> >   85eee811cdf644f759a2"..Content-Length: 0....
> >
> > U 2008/12/06 03:38:44.086313 196.212.209.18:46738 -> kamailio IP:5060
> >   ACK sip:0123825710 at kamailio IP SIP/2.0..Via: SIP/2.0/UDP
> > 192.168.0.55:467
> >   38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To:
> > "012382571
> >   0"<sip:0123825710 at kamailioIP
> >;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe.
> >   .From: <sip:1974 at 41.208.212.97 <sip%3A1974 at 41.208.212.97> <
> sip%3A1974 at 41.208.212.97 <sip%253A1974 at 41.208.212.97>
> >>;tag=353dd217..Call-ID:
> > MGRiNzM0ZGYxZTk1ZDI3
> >   ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 1 ACK..Content-Length: 0....
> >
> >
> >
> > U 2008/12/06 03:38:44.582208 asterisk IP:5065 -> kamailio IP:5060
> >   SIP/2.0 100 Trying..Via: SIP/2.0/UDP kamailio
> IP;branch=z9hG4bKd78.7bd576
> >   24.0;received=kamailio IP..Via: SIP/2.0/UDP 192.168.0.55:46738
> > ;received=1
> >   96.212.209.18
> > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=4673
> >   8..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From:
> <si
> >   p:1974 at kamailio IP>;tag=353dd217..To:
> > "0123825710"<sip:0123825710 at 41.208.
> >   212.97>..Call-ID: MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZjg...CSeq: 2
> > INV
> >   ITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS,
> BYE,
> > RE
> >   FER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact:
> > <sip:27123825710 at 41.2
> >   08.212.97:5065>..Content-Length: 0....
> >
> > U 2008/12/06 03:38:44.711225 70.42.72.49:5060 -> Asterisk IP:5065
> >   SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk
> IP:5065;branch=z9hG4b
> >   K22abec12;rport=5065..From: "1974" <sip:1974 at AsteriskIP
> :5065>;tag=as6a7c
> >   b89f..To: <sip:1214650027123825710 at 70.42.72.49<sip%3A1214650027123825710 at 70.42.72.49>
> <sip%3A1214650027123825710 at 70.42.72.49<sip%253A1214650027123825710 at 70.42.72.49>
> >>..Call-ID:
> > 1934f5d443abffe07
> >   c59d0a42215b49c at 41.208.212.97..CSeq: 102 INVITE..Server: OpenSER
> > (1.3.2-not
> >   ls (i386/solaris))..Content-Length: 0....
> >
> > U 2008/12/06 03:38:47.206445 70.42.72.49:5060 -> Asterisk IP:5065
> >   SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk
> IP:5065;branch=z9
> >   hG4bK22abec12;rport=5065..From: "1974" <sip:1974 at AsteriskIP
> :5065>;tag=as
> >   6a7cb89f..To: <sip:1214650027123825710 at 70.42.72.49<sip%3A1214650027123825710 at 70.42.72.49>
> <sip%3A1214650027123825710 at 70.42.72.49<sip%253A1214650027123825710 at 70.42.72.49>
> >
> > >;tag=cba-1a6e-48ecbab6..
> >   Call-ID: 1934f5d443abffe07c59d0a42215b49c at Asterisk IP..CSeq: 102
> > INVITE..
> >   Contact: <sip:1214650027123825710 at 70.42.72.138<sip%3A1214650027123825710 at 70.42.72.138>
> <sip%3A1214650027123825710 at 70.42.72.138<sip%253A1214650027123825710 at 70.42.72.138>
> >>..Date:
> > Wed, 08 Oct 2008 13:
> >   50:49 GMT..Server: BRSIP v2.0.1.2..Record-Route: <sip:70.42.72.49
> > ;lr=on;fta
> >   g=as6a7cb89f>..Content-Type: application/sdp..Content-Length:
> > 212....v=0..o
> >   =BRSDP 792898 792898 IN IP4 216.49.201.22..s=BRSDP Session..c=IN IP4
> > 216.49
> >   .201.22..t=0 0..m=audio 27852 RTP/AVP 0 101..a=ptime:20..a=rtpmap:0
> > PCMU/80
> >   00..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
> >
> > U 2008/12/06 03:38:47.206891 Asterisk IP:5065 -> Kamailio IP:5060
> >   SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP kamailio
> IP;branch=z9hG4bK
> >   d78.7bd57624.0;received=kamailio IP..Via: SIP/2.0/UDP
> 192.168.0.55:46738
> > ;
> >   received=196.212.209.18
> > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;
> >   rport=46738..Record-Route: <sip:41.208.212.97
> > ;lr=on;ftag=353dd217;nat=yes>.
> >   .From: <sip:1974 at kamilio IP>;tag=353dd217..To:
> "0123825710"<sip:01238257
> >   10 at 41.208.212.97>;tag=as4377a96d..Call-ID:
> > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVl
> >   NTQ0NTUwZjg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE,
> > ACK,
> >    CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported:
> > replaces..Conta
> >   ct: <sip:27123825710 at Asterisk IP:5065>..Content-Type:
> > application/sdp..Co
> >   ntent-Length: 287....v=0..o=root 2664 2664 IN IP4
> > 41.208.212.97..s=session.
> >  .c=IN IP4 41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3
> > 101..a=rtpmap:8
> >    PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101
> > telepho
> >   ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
> > -..a=ptime:20..a=se
> >   ndrecv..
> >
> > U 2008/12/06 03:38:47.207106 Kamailio IP:5060 -> 196.212.209.18:46738
> >   SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.55:46738
> > ;received=
> >   196.212.209.18
> > ;branch=z9hG4bK-d8754z-2aedec27a8ddd96c-1---d8754z-;rport=467
> >   38..Record-Route: <sip:kamailio IP;lr=on;ftag=353dd217;nat=yes>..From:
> <s
> >   ip:1974 at Kamailio IP>;tag=353dd217..To:
> > "0123825710"<sip:0123825710 at kamilio IP>;tag=as4377a96d..Call-ID:
> > MGRiNzM0ZGYxZTk1ZDI3ZjZmMTRhMDVlNTQ0NTUwZ
> >   jg...CSeq: 2 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
> > CANCEL,
> >   OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Contact:
> > <sip:
> >   27123825710 at 41.208.212.97:5065>..Content-Type:
> > application/sdp..Content-Len
> >   gth: 287....v=0..o=root 2664 2664 IN IP4 Asterisk IP..s=session..c=IN
> IP4
> >    41.208.212.97..t=0 0..m=audio 19202 RTP/AVP 8 0 3 101..a=rtpmap:8
> > PCMA/800
> >   0..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101
> > telephone-event/
> >   8000..a=fmtp:101 0-16..a=silenceSupp:off - - -
> > -..a=ptime:20..a=sendrecv..
> >
> >
> >
> ########################################################################################
> > Sip.conf
> > [general]
> > context=from-trunk
> > bindport=5065
> > autocreatepeer=yes
> > bindaddr=xx.xxx.xxx.xx
> >
> >
> > disallow=all
> > ;allow=gsm
> > ;allow=amr
> > allow=alaw
> > allow=ulaw
> > allow=gsm
> > ;allow=ilbc
> > ;disallow=all
> > ;
> > ;useragent=Asterisk PBX
> > dtmfmode = rfc2833
> >
> >
> > domain=xx.xxx.xxx.xx                   ; Add IP address as local domain
> >
> > [Provider]
> > disallow=all
> > canreinvite=no
> > context=from-trunk
> > allow=all
> > ;allow=ulaw
> > ;allow=gsm
> > host=xx.xx.xx.xx
> > insecure=port,invite
> > type=peer                               ; we only want to call out, not
> be
> > call$
> > dtmfmode=rfc2833
> >
> >
> #########################################################################################
> > Here is my codecs
> >
> > 41*CLI> core show translation
> >          Translation times between formats (in milliseconds) for one
> second
> > of data
> >           Source Format (Rows) Destination Format (Columns)
> >
> >           g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc
> g726
> > g722 amr
> >      g723    -   -    -    -        -     -    -     -    -     -    -
>  -
> >    -   -
> >       gsm    -   -    2    2        2     2    1     5    -    19    -
>  2
> >    -  14
> >      ulaw    -   5    -    1        2     2    1     5    -    19    -
>  2
> >    -  14
> >      alaw    -   5    1    -        2     2    1     5    -    19    -
>  2
> >    -  14
> >  g726aal2    -   5    2    2        -     2    1     5    -    19    -
>  1
> >    -  14
> >     adpcm    -   5    2    2        2     -    1     5    -    19    -
>  2
> >    -  14
> >      slin    -   4    1    1        1     1    -     4    -    18    -
>  1
> >    -  13
> >     lpc10    -   6    3    3        3     3    2     -    -    20    -
>  3
> >    -  15
> >      g729    -   -    -    -        -     -    -     -    -     -    -
>  -
> >    -   -
> >     speex    -   6    3    3        3     3    2     6    -     -    -
>  3
> >    -  15
> >      ilbc    -   -    -    -        -     -    -     -    -     -    -
>  -
> >    -   -
> >      g726    -   5    2    2        1     2    1     5    -    19    -
>  -
> >    -  14
> >      g722    -   -    -    -        -     -    -     -    -     -    -
>  -
> >    -   -
> >       amr    -   6    3    3        3     3    2     6    -    20    -
>  3
> >    -   -
> >
> >
> >
> > With best regards,
> > Lu.
> >
> > --
> > Luzango Mfupe
> > TUUNE MOBILE
> > Tel:0128440528/0123825710
> > Tshwane-RSA
> >
> > "...Ships are safe in harbor, but they were never meant to stay
> > there......."
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.kamailio.org
> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> >
> >
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> End of Users Digest, Vol 41, Issue 24
> *************************************
>



-- 
Luzango Mfupe
TUUNE MOBILE
Tel:0128440528/0123825710
Tshwane-RSA

"...Ships are safe in harbor, but they were never meant to stay
there......."
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