thanks Samy i have set nat=yes, now getting sound from both side but there is too uch disturbance. soetime we becoe audible and sometime not.i did not set extern ip coz my asterisk server is directly configured on public ip. I have softphones on some where localnets separate from asterisk server campus . i also set "sip set debug on" CLI prompt. this is giving following error.
when i test sip traffic on wireshark "401 unauthorize" error getting this error cli prompt also showing. my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone (9000) in another localnet in another campus(192.168.6.25) Scheduling destruction of SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' in 32000 ms (Method: INVITE) [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- SIP/9000-00000005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/9001-00000004' status is 'CONGESTION' <--- Reliably Transmitting (NAT) to 122.163.193.94:1801 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801 From: "9001"<sip:9001@122.160.154.189>;tag=b0785362 To: <sip:9000@122.160.154.189>;tag=as6c7d28d1 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 INVITE Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0 <------------> Really destroying SIP dialog ' 551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060' Method: INVITE <--- SIP read from UDP:122.163.193.94:1801 ---> ACK sip:9000@122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport Max-Forwards: 70 To: <sip:9000@122.160.154.189>;tag=as6c7d28d1 From: "9001"<sip:9001@122.160.154.189>;tag=b0785362 Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU. CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK <--- SIP read from UDP:122.163.193.94:1801 ---> <-------------> <--- SIP read from UDP:115.249.67.250:5060 ---> REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp Max-Forwards: 70 To: "shekhar" <sip:9000@122.160.154.189> From: "shekhar" <sip:9000@122.160.154.189>;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Contact: <sip:9000@192.168.6.25>;expires=3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) <--- Transmitting (NAT) to 115.249.67.250:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060 From: "shekhar" <sip:9000@122.160.154.189>;tag=jcysf To: "shekhar" <sip:9000@122.160.154.189>;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 954 REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="278a3764" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:115.249.67.250:5060 ---> REGISTER sip:122.160.154.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKjmqlllxn Max-Forwards: 70 To: "shekhar" <sip:9000@122.160.154.189> From: "shekhar" <sip:9000@122.160.154.189>;tag=jcysf Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955 REGISTER Contact: <sip:9000@192.168.6.25>;expires=3600 Authorization: Digest username="9000",realm="asterisk",nonce="278a3764",uri="sip:122.160.154.189",response="c7a119185514202d5f9cc10a86a93607",algorithm=MD5 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE User-Agent: Twinkle/1.4.2 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 115.249.67.250:5060 (NAT) <--- Transmitting (NAT) to 115.249.67.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKjmqlllxn;received=115.249.67.250;rport=5060 From: "shekhar" <sip:9000@122.160.154.189>;tag=jcysf To: "shekhar" <sip:9000@122.160.154.189>;tag=as26d4cd86 Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110 CSeq: 955 REGISTER Server: Asterisk PBX 10.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 3600 Contact: <sip:9000@192.168.6.25>;expires=3600 Date: Wed, 04 Jul 2012 14:08:17 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:122.163.193.94:1801 ---> <-------------> Really destroying SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' Method: REGISTER regards abhi On Mon, Jul 2, 2012 at 5:22 PM, SamyGo <govoi...@gmail.com> wrote: > actually its a one-way audio issue due to NAT ! > > alok , please explain your network flow for end to end > client-server-client. > > You may need to set nat=yes for your sip peer behind NAT. If the server is > behind NAT router/firewall use externip=<public.ip.of.server> field. > Also provide sip traces of this call. > Another thing to do for your learning. Execute wireshark on both softphone > systems and set "sip | rtp" as filter and see where are the RTP streams > going on each end ! > > Take a complete capture on Asterisk server by executing the command "sip > set debug on" and make a call. > > BR > Sammy > > > On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon > <dig...@sanguinarius.co.uk>wrote: > >> alok srivastava wrote: >> >>> dear >>> i have configured properly asterisk. At the one end i am using x-lite >>> soft ph and another end twinkle. call is going properly from both end but >>> after picking the phone not able to listen other one. >>> when i checked the port 5060 on the asterisk server it is always showing >>> closed while i have flushed all the rules from iptables (iptables -F) >>> >>> PORT STATE SERVICE VERSION >>> 5060/tcp closed sip >>> >>> telnet localhost 5060 (could not connect) >>> >>> regards >>> alok >>> >>> >>> SIP is only used to setup (and stop etc.) the call. The actual audio is >> sent via rtp. >> >> /etc/asterisk/rtp.conf >> >> Should tell which ports asterisk is using for rtp, you will need to make >> sure that the remote host can connect to these ports. >> >> There are lots of articles around on how to resolve this. >> >> >> >> >> -- >> ______________________________**______________________________**_________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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