[Kamailio-Users] I don't have asterisk audio to openser - mediaproxy

Fri Oct 31 04:16:39 CET 2008

I don't have problems when I make calls to the pstn I listen well and people listen to me well, the problem is when I receive a call from the pstn I don't listen anything and they don't listen to me, inside the sip.conf already has configured the values nat, externip localnet .

I believe that the problem is that openser detects as nat the ip of my asterisk, eye > "I have the openser and the mediaproxy with asterisk in the same pc"

### Sip Log Asterisk  #### 

<--- SIP read from 192.168.10.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070
Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes>
From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as4f7a434f
To: <sip:113 at 192.168.10.1>;tag=a72df908ec08f63d
Call-ID: 1661965d0c28650d517426333e4a6ae5 at 192.168.10.1
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:113 at 192.168.10.30:5062;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
P-hint: Onreply-route - fixcontact

<------------->
--- (12 headers 0 lines) ---
    -- SIP/openser-08c0ea58 is ringing
xserver*CLI>
<--- SIP read from 192.168.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK5839d960;rport=5070
Record-Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes>
From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as4f7a434f
To: <sip:113 at 192.168.10.1>;tag=a72df908ec08f63d
Call-ID: 1661965d0c28650d517426333e4a6ae5 at 192.168.10.1
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:113 at 192.168.10.30:5062;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 212
P-hint: Onreply-route - fixcontact
P-hint: onreply_route|usemediaproxy

v=0
o=113 8000 8000 IN IP4 192.168.10.30
s=SIP Call
c=IN IP4 192.168.1.64
t=0 0
m=audio 35004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (15 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.64:35004
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.64:35004
list_route: hop: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes>
set_destination: Parsing <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes> for address/port to send to
set_destination: set destination to 192.168.10.1, port 5060
Transmitting (NAT) to 192.168.10.1:5060:
ACK sip:113 at 192.168.10.30:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5070;branch=z9hG4bK32b6019c;rport
Route: <sip:192.168.10.1;lr=on;ftag=as4f7a434f;nat=yes>
From: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as4f7a434f
To: <sip:113 at 192.168.10.1>;tag=a72df908ec08f63d
Contact: <sip:asterisk at 192.168.10.1:5070>
Call-ID: 1661965d0c28650d517426333e4a6ae5 at 192.168.10.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
    -- SIP/openser-08c0ea58 answered Zap/4-1
xserver*CLI>
<--- SIP read from 192.168.10.1:5060 --->
BYE sip:asterisk at 192.168.10.1:5070 SIP/2.0
Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0
Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0
From: <sip:113 at 192.168.10.1>;tag=a72df908ec08f63d
To: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as4f7a434f
Supported: path
Call-ID: 1661965d0c28650d517426333e4a6ae5 at 192.168.10.1
CSeq: 9793 BYE
User-Agent: Grandstream GXP2020 1.1.6.16
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
P-hint: LR|fixcontact,setflag6

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.10.1 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.10.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK71b2.32123901.0;received=192.168.10.1
Via: SIP/2.0/UDP 192.168.10.30:5062;branch=z9hG4bK02603cb0e798dac0
Record-Route: <sip:192.168.10.1;lr=on;ftag=a72df908ec08f63d>
From: <sip:113 at 192.168.10.1>;tag=a72df908ec08f63d
To: "asterisk" <sip:asterisk at 192.168.10.1:5070>;tag=as4f7a434f
Call-ID: 1661965d0c28650d517426333e4a6ae5 at 192.168.10.1
CSeq: 9793 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:asterisk at 192.168.10.1:5070>
Content-Length: 0

________________________________
From: luzango mfupe <luzango.mfupe at gmail.com>

Hi RickyI should have seen how you handle NAT in kamaiilo.conf but you can also edit sip.conf in Asterisk and  try to put Nat=yes
Rgds,

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