Ok David, will give feedback soon.rgds, Lu.
On Thu, Oct 9, 2008 at 11:53 AM, users-request@lists.kamailio.org wrote:
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Today's Topics:
- Re: No audio both ways. (David Villasmil)
Message: 1 Date: Thu, 9 Oct 2008 11:07:12 +0200 From: "David Villasmil" david.villasmil.work@gmail.com Subject: Re: [Kamailio-Users] No audio both ways. Cc: users@lists.kamailio.org Message-ID: 9853f4ff0810090207p11d15bddt8dca8f57370c83c9@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@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]+@.*") { 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]+@.*") { 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@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@196.212.209.18:46738..To: "0123825710"<sip:01238 25710@kamailio IP>..From: sip:1974@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@kamailioIP ;tag=329cfea a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@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@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@kamailioIP ;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe. .From: <sip:1974@41.208.212.97 sip%3A1974@41.208.212.97 <
sip%3A1974@41.208.212.97 sip%253A1974@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@kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@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@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@AsteriskIP
:5065>;tag=as6a7c
b89f..To: <sip:1214650027123825710@70.42.72.49sip%3A1214650027123825710@70.42.72.49
<sip%3A1214650027123825710@70.42.72.49sip%253A1214650027123825710@70.42.72.49
..Call-ID:
1934f5d443abffe07 c59d0a42215b49c@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@AsteriskIP
:5065>;tag=as
6a7cb89f..To: <sip:1214650027123825710@70.42.72.49sip%3A1214650027123825710@70.42.72.49
<sip%3A1214650027123825710@70.42.72.49sip%253A1214650027123825710@70.42.72.49
;tag=cba-1a6e-48ecbab6..
Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 INVITE.. Contact: <sip:1214650027123825710@70.42.72.138sip%3A1214650027123825710@70.42.72.138
<sip%3A1214650027123825710@70.42.72.138sip%253A1214650027123825710@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@kamilio IP>;tag=353dd217..To:
"0123825710"<sip:01238257
10@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@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@Kamailio IP>;tag=353dd217..To: "0123825710"<sip:0123825710@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@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 ilbcg726
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......."
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luzango mfupe