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;
};
};
###################################################################################
This call was from the xlite softphone 1974 towards the Landline 0123825710.
###################################################################################
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
25710@kamailio IP>..From: <sip:1974@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
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@kamailio IP>;tag=329cfea
a6ded039da25ff8cbb8668bd2.dcfe..From: <sip:1974@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....
38;branch=z9hG4bK-d8754z-2d5e73428b95d262-1---d8754z-;rport..To: "012382571
0"<sip:0123825710@kamailio IP>;tag=329cfeaa6ded039da25ff8cbb8668bd2.dcfe.
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....
SIP/2.0 100 Giving a try..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9hG4b
K22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as6a7c
c59d0a42215b49c@41.208.212.97..CSeq: 102 INVITE..Server: OpenSER (1.3.2-not
ls (i386/solaris))..Content-Length: 0....
SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Asterisk IP:5065;branch=z9
hG4bK22abec12;rport=5065..From: "1974" <sip:1974@Asterisk IP:5065>;tag=as
Call-ID: 1934f5d443abffe07c59d0a42215b49c@Asterisk IP..CSeq: 102 INVITE..
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
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
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..
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:
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.