Hi all,<br clear="all"><br>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;<br>
<br>localnet=<a href="http://130.0.0.0/130.0.0.0">130.0.0.0/130.0.0.0</a><br>externhost=12.131.12.13<br>externrefresh=10<br>fromdomain=<a href="http://test.localhost.com">test.localhost.com</a><br>nat=yes<br>qualify=yes<br>
canreinvite=no<br><br><br>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;<br><br>Reliably Transmitting (NAT) to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a>:<br>
INVITE <a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a> SIP/2.0<br>Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK1fbbab95;rport<br>Max-Forwards: 70<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>><br>Contact: <<a href="http://sip:77057@12.131.12.13:5060">sip:77057@12.131.12.13:5060</a>><br>Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>
CSeq: 102 INVITE<br>User-Agent: FPBX-2.9.0(1.8.5.0)<br>Date: Wed, 01 Feb 2012 16:11:35 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 285<br><br>v=0<br>o=root 122642112 122642112 IN IP4 12.131.12.13<br>s=Asterisk PBX 1.8.5.0<br>c=IN IP4 12.131.12.13<br>t=0 0<br>m=audio 16006 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br><br>---<br> -- Called SIP/ATTLABS-IP-FlexReach/173242<br><br><--- SIP read from UDP:<a href="http://12.194.12.12:5060">12.194.12.12:5060</a> ---><br>
SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 12.131.12.13:5060;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>><br>Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>
CSeq: 102 INVITE<br><br><-------------><br>--- (6 headers 0 lines) ---<br><br><--- SIP read from UDP:<a href="http://12.194.12.12:5060">12.194.12.12:5060</a> ---><br>SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 12.131.12.13:5060;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060<br>
From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400<br>
Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>CSeq: 102 INVITE<br>Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK<br>Contact: <sip:12.194.12.12:5060;transport=udp><br>
Content-Length: 237<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br><br>v=0<br>o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12<br>s=SIP Media Capabilities<br>c=IN IP4 12.194.12.12<br>t=0 0<br>
m=audio 16534 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><-------------><br>--- (11 headers 11 lines) ---<br>Found RTP audio format 0<br>
Found RTP audio format 101<br>Found audio description format PCMU for ID 0<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)<br>Peer audio RTP is at port <a href="http://12.194.12.12:16534">12.194.12.12:16534</a><br> -- SIP/ATTLABS-IP-FlexReach-00000025 is making progress passing it to SIP/2005-00000024<br>
<br><--- SIP read from UDP:<a href="http://12.194.12.12:5060">12.194.12.12:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 12.131.12.13:5060;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400<br>Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>
CSeq: 102 INVITE<br>Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK<br>Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed<br>Contact: <sip:12.194.12.12:5060;transport=udp><br>
Content-Length: 237<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br><br>v=0<br>o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12<br>s=SIP Media Capabilities<br>c=IN IP4 12.194.12.12<br>t=0 0<br>
m=audio 16534 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><-------------><br>--- (12 headers 11 lines) ---<br>list_route: hop: <sip:12.194.12.12:5060;transport=udp><br>
set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for address/port to send to<br>set_destination: set destination to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a><br>Transmitting (NAT) to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a>:<br>
ACK sip:12.194.12.12:5060;transport=udp SIP/2.0<br>Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK483f052d;rport<br>Max-Forwards: 70<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400<br>Contact: <<a href="http://sip:77057@12.131.12.13:5060">sip:77057@12.131.12.13:5060</a>><br>
Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>CSeq: 102 ACK<br>User-Agent: FPBX-2.9.0(1.8.5.0)<br>Content-Length: 0<br><br><br>---<br>
-- SIP/ATTLABS-IP-FlexReach-00000025 answered SIP/2005-00000024<br> -- Locally bridging SIP/2005-00000024 and SIP/ATTLABS-IP-FlexReach-00000025<br>Reliably Transmitting (NAT) to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a>:<br>
OPTIONS sip:12.194.12.12 SIP/2.0<br>Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK06532068;rport<br>Max-Forwards: 70<br>From: "Unknown" <<a href="mailto:sip%3AUnknown@test.localhost.com">sip:Unknown@test.localhost.com</a>>;tag=as054a7d2d<br>
To: <sip:12.194.12.12><br>Contact: <<a href="http://sip:Unknown@12.131.12.13:5060">sip:Unknown@12.131.12.13:5060</a>><br>Call-ID: <a href="mailto:767dcb7d4406d06c248a7056559ad301@test.localhost.com">767dcb7d4406d06c248a7056559ad301@test.localhost.com</a><br>
CSeq: 102 OPTIONS<br>User-Agent: FPBX-2.9.0(1.8.5.0)<br>Date: Wed, 01 Feb 2012 16:11:46 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Length: 0<br>
<br><br>---<br><br><--- SIP read from UDP:<a href="http://12.194.12.12:5060">12.194.12.12:5060</a> ---><br>SIP/2.0 405 Method Not Allowed<br>Via: SIP/2.0/UDP 12.131.12.13:5060;received=12.131.12.13;branch=z9hG4bK06532068;rport=5060<br>
From: "Unknown" <<a href="mailto:sip%3AUnknown@test.localhost.com">sip:Unknown@test.localhost.com</a>>;tag=as054a7d2d<br>To: <sip:12.194.12.12>;tag=aprqngfrt-d1v40r10000c6<br>Call-ID: <a href="mailto:767dcb7d4406d06c248a7056559ad301@test.localhost.com">767dcb7d4406d06c248a7056559ad301@test.localhost.com</a><br>
CSeq: 102 OPTIONS<br>Reason: Q.850;cause=55;text="Call Terminated"<br>Allow: INVITE,ACK,BYE,CANCEL,PRACK,INFO,REFER,UPDATE,MESSAGE,PUBLISH<br><br><-------------><br>--- (8 headers 0 lines) ---<br>Scheduling destruction of SIP dialog '<a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a>' in 6400 ms (Method: INVITE)<br>
set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for address/port to send to<br>set_destination: set destination to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a><br>Reliably Transmitting (NAT) to <a href="http://12.194.12.12:5060">12.194.12.12:5060</a>:<br>
BYE sip:12.194.12.12:5060;transport=udp SIP/2.0<br>Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK2ab85b31;rport<br>Max-Forwards: 70<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400<br>Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>
CSeq: 103 BYE<br>User-Agent: FPBX-2.9.0(1.8.5.0)<br>X-Asterisk-HangupCause: Normal Clearing<br>X-Asterisk-HangupCauseCode: 16<br>Content-Length: 0<br><br><br>---<br><--- SIP read from UDP:<a href="http://12.194.12.12:5060">12.194.12.12:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 12.131.12.13:5060;received=12.131.12.13;branch=z9hG4bK2ab85b31;rport=5060<br>From: "77057" <<a href="mailto:sip%3A77057@test.localhost.com">sip:77057@test.localhost.com</a>>;tag=as1fa9b502<br>
To: <<a href="mailto:sip%3A173242@12.194.12.12">sip:173242@12.194.12.12</a>>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400<br>Call-ID: <a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a><br>
CSeq: 103 BYE<br>Content-Length: 0<br><br><-------------><br>--- (7 headers 0 lines) ---<br>Really destroying SIP dialog '<a href="mailto:04ce1d566f1f17a221caba261e2af4bb@test.localhost.com">04ce1d566f1f17a221caba261e2af4bb@test.localhost.com</a>' Method: INVITE<br>
<br><br>The Asterisk version I'm using is 1.8.5. Please assist me at earliest.<br><br>-- <br>Regards,<br><br>Ahmed Munir Chohan<br><br><br>