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@192.168.10.1:5070;tag=as4f7a434f To: sip:113@192.168.10.1;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@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@192.168.10.1:5070;tag=as4f7a434f To: sip:113@192.168.10.1;tag=a72df908ec08f63d Call-ID: 1661965d0c28650d517426333e4a6ae5@192.168.10.1 CSeq: 102 INVITE User-Agent: Grandstream GXP2020 1.1.6.16 Contact: sip:113@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@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@192.168.10.1:5070;tag=as4f7a434f To: sip:113@192.168.10.1;tag=a72df908ec08f63d Contact: sip:asterisk@192.168.10.1:5070 Call-ID: 1661965d0c28650d517426333e4a6ae5@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@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@192.168.10.1;tag=a72df908ec08f63d To: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as4f7a434f Supported: path Call-ID: 1661965d0c28650d517426333e4a6ae5@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@192.168.10.1;tag=a72df908ec08f63d To: "asterisk" sip:asterisk@192.168.10.1:5070;tag=as4f7a434f Call-ID: 1661965d0c28650d517426333e4a6ae5@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@192.168.10.1:5070 Content-Length: 0
________________________________ From: luzango mfupe luzango.mfupe@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,