thanks Jim I will check stun server settings asap,
but i have noticed 192.168.x.x is also present in the debug of successful call having both way audio. so i don't think this has to do anything with this. below is the sip debug of successful call . --- Audio is at 79.80.154.99 port 14034 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 116.18.35.235:28614: INVITE sip:[email protected]:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport From: "pepsi coke" <sip:[email protected]:5678>;tag=as12245807 To: <sip:[email protected]:28614;rinstance=0266b8b94f488588> Contact: <sip:[email protected]:5678> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:06:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 9626 9626 IN IP4 79.80.154.99 s=session c=IN IP4 79.80.154.99 t=0 0 m=audio 14034 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 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 --- [Jul 21 11:06:24] WARNING[23749]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called adf ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678 Contact: <sip:[email protected]:28614;rinstance=0266b8b94f488588> To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=bd6f2350 From: "pepsi coke"<sip:[email protected]:5678>;tag=as12245807 Call-ID: [email protected] CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/adf-00794e30 is ringing ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678 Contact: <sip:[email protected]:28614;rinstance=0266b8b94f488588> To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=bd6f2350 From: "pepsi coke"<sip:[email protected]:5678>;tag=as12245807 Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 185 v=0 o=- 2 2 IN IP4 192.168.0.12 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.12 t=0 0 m=audio 15956 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.12:15956 Found description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.12:15956 list_route: hop: <sip:[email protected]:28614;rinstance=0266b8b94f488588> [Jul 21 11:06:38] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session [email protected] set_destination: Parsing <sip:[email protected]:28614;rinstance=0266b8b94f488588> for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Transmitting (NAT) to 116.18.35.235:28614: ACK sip:[email protected]:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK00fdcc7c;rport From: "pepsi coke" <sip:[email protected]:5678>;tag=as12245807 To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=bd6f2350 Contact: <sip:[email protected]:5678> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/adf-00794e30 left from hold -- SIP/adf-00794e30 answered SIP/pepsi-9fd06cc0 ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> <-------------> --- (0 headers 1 lines) --- ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SUBSCRIBE sip:[email protected]:5678 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:28614 ;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:[email protected]:28614> To: "adf"<sip:[email protected]:5678> From: "adf"<sip:[email protected]:5678>;tag=5d297f22 Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Event: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Creating new subscription Sending to 116.18.35.235 : 28614 (NAT) Found peer 'adf' Looking for adf in uscan_int (domain ast-server.axvoice.com) ast-server*CLI> <--- Transmitting (NAT) to 116.18.35.235:28614 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.12:28614 ;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;received=116.18.35.235;rport=28614 From: "adf"<sip:[email protected]:5678>;tag=5d297f22 To: "adf"<sip:[email protected]:5678>;tag=as724c598c Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY. CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Really destroying SIP dialog 'MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.' Method: SUBSCRIBE Reliably Transmitting (NAT) to 116.18.35.235:28614: OPTIONS sip:[email protected]:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport From: "asterisk" <sip:[email protected]:5678>;tag=as223ef4a7 To: <sip:[email protected]:28614;rinstance=0266b8b94f488588> Contact: <sip:[email protected]:5678> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:07:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport=5678 Contact: <sip:192.168.0.12:28614> To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=15133f38 From: "asterisk"<sip:[email protected]:5678>;tag=as223ef4a7 Call-ID: [email protected] CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS ast-server*CLI> <------------> Scheduling destruction of SIP dialog ' [email protected]' in 23936 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 116.18.35.235:28614: NOTIFY sip:[email protected]:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport From: "asterisk" <sip:[email protected]:5678>;tag=as756cae64 To: <sip:[email protected]:28614;rinstance=0266b8b94f488588> Contact: <sip:[email protected]:5678> Call-ID: [email protected] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:[email protected] <sip%[email protected]> Voice-Message: 0/0 (0/0) --- ast-server*CLI> <--- SIP read from 116.18.35.235:28614 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport=5678 Contact: <sip:192.168.0.12:28614> To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=b9541904 From: "asterisk"<sip:[email protected]:5678>;tag=as756cae64 Call-ID: [email protected] CSeq: 102 NOTIFY User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: NOTIFY [Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:3074 update_call_counter: Call to peer 'adf' removed from call limit 2 Scheduling destruction of SIP dialog ' [email protected]' in 18624 ms (Method: INVITE) [Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:5695 reqprep: Strict routing enforced for session [email protected] set_destination: Parsing <sip:[email protected]:28614;rinstance=0266b8b94f488588> for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Reliably Transmitting (NAT) to 116.18.35.235:28614: BYE sip:[email protected]:28614;rinstance=0266b8b94f488588 SIP/2.0 Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK05cc42e6;rport From: "pepsi coke" <sip:[email protected]:5678>;tag=as12245807 To: <sip:[email protected]:28614;rinstance=0266b8b94f488588>;tag=bd6f2350 Call-ID: [email protected] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 >Date: Wed, 28 Jul 2010 09:36:51 -0700 >From: Jim Dickenson <[email protected]> >Subject: Re: [asterisk-users] Nat issue one way audio on IP dial >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> >Message-ID: <[email protected]> >Content-Type: text/plain; charset="us-ascii" >Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP >debug output a 192.168 address for the RTP packets to go to which of course will not work. >-- >Jim Dickenson >mailto:[email protected] <[email protected]> >CfMC >http://www.cfmc.com/
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