[Kamailio-Users] No audio both ways.

David Villasmil david.villasmil.work at gmail.com
Thu Oct 9 11:07:12 CEST 2008


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 kamailio IP>;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>>;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>>..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>
> >;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>>..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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