[Kamailio-Users] No audio both ways.
luzango mfupe
luzango.mfupe at gmail.com
Thu Oct 9 10:58:36 CEST 2008
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 kamailio IP>;tag=329cfea
a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974 at kamailio IP>;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 kamailio IP>;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 Asterisk IP: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 Asterisk IP: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 kamilioIP>;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......."
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