subject:"\[asterisk\-users\] port 5060 is blocked by ISP"

Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-04 Thread SamyGo

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  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: "9001";tag=b0785362
> To: ;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: ;tag=as6c7d28d1
> From: "9001";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" 
> From: "shekhar" ;tag=jcysf
> Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110
> CSeq: 954 REGISTER
> Contact: ;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" ;tag=jcysf
> To: "shekhar" ;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 'qajoim

Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-04 Thread alok srivastava

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: "9001";tag=b0785362
To: ;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: ;tag=as6c7d28d1
From: "9001";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" 
From: "shekhar" ;tag=jcysf
Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110
CSeq: 954 REGISTER
Contact: ;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" ;tag=jcysf
To: "shekhar" ;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" 
From: "shekhar" ;tag=jcysf
Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110
CSeq: 955 REGISTER
Contact: ;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" ;tag=jcysf
To: "shekhar" ;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: ;expires=3600
Date: Wed, 04 Jul 2012 14:08:17 GMT
Content-Length: 0


<>
Scheduling destruct

Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-02 Thread SamyGo

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= 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 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.
>
>
>
>
> --
> __**__**_
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>   
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Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-02 Thread Thomas Kenyon


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.



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Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-02 Thread Olle E. Johansson


1 jul 2012 kl. 09:48 skrev Leandro Dardini:

> Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet 
> is using TCP.

That's not correct. SIP supports multiple transports, including TCP. Not all 
implementations support TCP though.

/O
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Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-01 Thread Hans Witvliet

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


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Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-01 Thread B. van Ouwerkerk


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



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Re: [asterisk-users] port 5060 is blocked by ISP

2012-07-01 Thread Leandro Dardini

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"  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 --
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>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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[asterisk-users] port 5060 is blocked by ISP

2012-07-01 Thread alok srivastava

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
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Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

Re: [asterisk-users] port 5060 is blocked by ISP

[asterisk-users] port 5060 is blocked by ISP

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