Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below;
localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13 externrefresh=10 fromdomain=test.localhost.com nat=yes qualify=yes canreinvite=no NAT on device end i.e. my softphone (extension) has already set to yes with canreinvite=no but still unable to resolve this issue. SIP traces are listed below; Reliably Transmitting (NAT) to 12.194.12.12:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK1fbbab95;rport Max-Forwards: 70 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: FPBX-2.9.0(1.8.5.0) Date: Wed, 01 Feb 2012 16:11:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 v=0 o=root 122642112 122642112 IN IP4 12.131.12.13 s=Asterisk PBX 1.8.5.0 c=IN IP4 12.131.12.13 t=0 0 m=audio 16006 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called SIP/ATTLABS-IP-FlexReach/173242 <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected] >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK Contact: <sip:12.194.12.12:5060;transport=udp> Content-Length: 237 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12 s=SIP Media Capabilities c=IN IP4 12.194.12.12 t=0 0 m=audio 16534 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=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 12.194.12.12:16534 -- SIP/ATTLABS-IP-FlexReach-00000025 is making progress passing it to SIP/2005-00000024 <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected] >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:12.194.12.12:5060;transport=udp> Content-Length: 237 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12 s=SIP Media Capabilities c=IN IP4 12.194.12.12 t=0 0 m=audio 16534 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 <-------------> --- (12 headers 11 lines) --- list_route: hop: <sip:12.194.12.12:5060;transport=udp> set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for address/port to send to set_destination: set destination to 12.194.12.12:5060 Transmitting (NAT) to 12.194.12.12:5060: ACK sip:12.194.12.12:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK483f052d;rport Max-Forwards: 70 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected] >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Contact: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.5.0) Content-Length: 0 --- -- SIP/ATTLABS-IP-FlexReach-00000025 answered SIP/2005-00000024 -- Locally bridging SIP/2005-00000024 and SIP/ATTLABS-IP-FlexReach-00000025 Reliably Transmitting (NAT) to 12.194.12.12:5060: OPTIONS sip:12.194.12.12 SIP/2.0 Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK06532068;rport Max-Forwards: 70 From: "Unknown" <sip:[email protected]>;tag=as054a7d2d To: <sip:12.194.12.12> Contact: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: FPBX-2.9.0(1.8.5.0) Date: Wed, 01 Feb 2012 16:11:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK06532068;rport=5060 From: "Unknown" <sip:[email protected]>;tag=as054a7d2d To: <sip:12.194.12.12>;tag=aprqngfrt-d1v40r10000c6 Call-ID: [email protected] CSeq: 102 OPTIONS Reason: Q.850;cause=55;text="Call Terminated" Allow: INVITE,ACK,BYE,CANCEL,PRACK,INFO,REFER,UPDATE,MESSAGE,PUBLISH <-------------> --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog ' [email protected]' in 6400 ms (Method: INVITE) set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for address/port to send to set_destination: set destination to 12.194.12.12:5060 Reliably Transmitting (NAT) to 12.194.12.12:5060: BYE sip:12.194.12.12:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK2ab85b31;rport Max-Forwards: 70 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected] >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Call-ID: [email protected] CSeq: 103 BYE User-Agent: FPBX-2.9.0(1.8.5.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:12.194.12.12:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 12.131.12.13:5060 ;received=12.131.12.13;branch=z9hG4bK2ab85b31;rport=5060 From: "77057" <sip:[email protected]>;tag=as1fa9b502 To: <sip:[email protected] >;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400 Call-ID: [email protected] CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog ' [email protected]' Method: INVITE The Asterisk version I'm using is 1.8.5. Please assist me at earliest. -- Regards, Ahmed Munir Chohan
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