Re: [asterisk-users] port 5060 is blocked by ISP
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/9000-0005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/9001-0004' 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: 9001sip: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: 9001sip: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
Re: [asterisk-users] port 5060 is blocked by ISP
Hi, Being audible sometime or bad voice quality is only due to internet latency or bad internet situation. [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). The above lines again telling that there is some problem sending sequential packets to some endpoint. That may lead to disconnection of call after some time..as it is currently doing so. Try setting some more NAT parameters...as you said localnet. Set *localnet=*parameter entries in your asterisk server sip configurations. BR Sammy On Wed, Jul 4, 2012 at 2:13 PM, alok srivastava alok...@gmail.com wrote: 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/9000-0005 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/9001-0004' 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: 9001sip: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: 9001sip: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,
Re: [asterisk-users] port 5060 is blocked by ISP
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet is using TCP. I am typing from my mobile phone... Il giorno 01/lug/2012 09:35, alok srivastava alok...@gmail.com ha scritto: 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] port 5060 is blocked by ISP
No voice means you have to look at the rtp ports. You can find more via google firewall rtp ports asterisk B. Op 1-7-2012 9:34, alok srivastava schreef: 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] port 5060 is blocked by ISP
On Sun, 2012-07-01 at 13:04 +0530, 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 Hi Alok, telnet is a very crude tool to test with. Try hping or nmap instead. Hans -- _ -- 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
