[asterisk-users] port 5060 is blocked by ISP
alok srivastava
alokkic at gmail.com
Wed Jul 4 04:13:10 CDT 2012
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 at 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 at 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 at 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 at 122.160.154.189>;tag=b0785362
To: <sip:9000 at 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 at 122.160.154.189:5060' Method: INVITE
<--- SIP read from UDP:122.163.193.94:1801 --->
ACK sip:9000 at 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 at 122.160.154.189>;tag=as6c7d28d1
From: "9001"<sip:9001 at 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 at 122.160.154.189>
From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
CSeq: 954 REGISTER
Contact: <sip:9000 at 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 at 122.160.154.189>;tag=jcysf
To: "shekhar" <sip:9000 at 122.160.154.189>;tag=as26d4cd86
Call-ID: qajoimvsihxpsff at 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 at 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 at 122.160.154.189>
From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
CSeq: 955 REGISTER
Contact: <sip:9000 at 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 at 122.160.154.189>;tag=jcysf
To: "shekhar" <sip:9000 at 122.160.154.189>;tag=as26d4cd86
Call-ID: qajoimvsihxpsff at 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 at 192.168.6.25>;expires=3600
Date: Wed, 04 Jul 2012 14:08:17 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'qajoimvsihxpsff at alok-Inspiron-N5110'
in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:122.163.193.94:1801 --->
<------------->
Really destroying SIP dialog 'qajoimvsihxpsff at alok-Inspiron-N5110' Method:
REGISTER
regards
abhi
On Mon, Jul 2, 2012 at 5:22 PM, SamyGo <govoiper at 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 <digium at 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.
>>
>>
>>
>>
>> --
>> ______________________________**______________________________**_________
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>
>
> --
> _____________________________________________________________________
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