Asterisk,SIP Retransmission timeout [closed]

F

5

10

I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok.
But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk 'B' and After 38 seconds call has been disconnected showing following warnings :

Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

Any ideas ?

Fussell answered 28/2, 2014 at 11:3 Comment(5)

can you call the DDI on your server 'A' without the trunk (A-B) configuation? To make sure your dialplan is correct. – Frederiksberg 28/2, 2014 at 12:47

Yes, Its working perfectly without trunk configuration. The main issue is call is being disconnected after 38 seconds, Before 38 seconds I can listen audio files to, As soon as the call reaches to 38th seconds , I t dropped – Fussell 28/2, 2014 at 13:45

so...have you solved your problem yet? – Mcduffie 20/11, 2014 at 8:40

Yes, As I mentioned it below it was due to nat setting in sip.conf – Fussell 4/12, 2014 at 11:52

please tag it as solved ;) thanks – Kurtzig 9/12, 2014 at 8:51

R

6

Such situation can be spot when you have nat issues or firewall issue

See this article http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

For more info you can enable sip debug by using

 asterisk -r
 sip set debug on

Rosamondrosamund answered 28/2, 2014 at 14:4 Comment(6)

I have removed the firewall settings, This issue is not related firewall, For Nat I have used nat=force_rport,comedia in sip.conf. – Fussell 28/2, 2014 at 14:25

From debug log, Asterisk server 'B' keeps on showing Retransmitting #{1-9}(NAT), where {1-9} showing retransmition attempt, – Fussell 1/3, 2014 at 7:7

Retransmitting #3 (NAT) to 111.XXX.XXX.X:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.XXX.XXX.X:5060;branch=z9hG4bK5e38e44b;received=111.XXX.XXX.X;rport=5060 From: <sip:[email protected]>;tag=as086c55da To: <sip:[email protected]>;tag=as36de7a5b Call-ID: [email protected]:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp – Fussell 1/3, 2014 at 7:8

So you have nat or firewall issues like i say. – Rosamondrosamund 1/3, 2014 at 16:33

Thanks for your reaponse, please suggest me the solution to overcome this problem – Fussell 2/3, 2014 at 17:20

Setup nat/fix firewall. – Rosamondrosamund 2/3, 2014 at 20:15

Z

10

By default Asterisk sends a RE-INVITE request after a call is established.

But most sip clients and sip servers in the market do not accept RE-INVITE requests. For this reason, when Asterisk sends a RE-INVITE after a call is established, the other side does not answer the request. So, after 32 seconds, Asterisk hangs up the call.

To solve the problem, you need to disable the RE-INVITE feature of Asterisk if your client software does not accept RE-INVITE requests. To do this, you need to edit the sip.conf file in Asterisk to include:

canreinvite = no

Zooplankton answered 5/5, 2015 at 16:39 Comment(0)

R

6

Such situation can be spot when you have nat issues or firewall issue

See this article http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions

For more info you can enable sip debug by using

 asterisk -r
 sip set debug on

Rosamondrosamund answered 28/2, 2014 at 14:4 Comment(6)

I have removed the firewall settings, This issue is not related firewall, For Nat I have used nat=force_rport,comedia in sip.conf. – Fussell 28/2, 2014 at 14:25

From debug log, Asterisk server 'B' keeps on showing Retransmitting #{1-9}(NAT), where {1-9} showing retransmition attempt, – Fussell 1/3, 2014 at 7:7

Retransmitting #3 (NAT) to 111.XXX.XXX.X:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.XXX.XXX.X:5060;branch=z9hG4bK5e38e44b;received=111.XXX.XXX.X;rport=5060 From: <sip:[email protected]>;tag=as086c55da To: <sip:[email protected]>;tag=as36de7a5b Call-ID: [email protected]:5060 CSeq: 102 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:[email protected]:5060> Content-Type: application/sdp – Fussell 1/3, 2014 at 7:8

So you have nat or firewall issues like i say. – Rosamondrosamund 1/3, 2014 at 16:33

Thanks for your reaponse, please suggest me the solution to overcome this problem – Fussell 2/3, 2014 at 17:20

Setup nat/fix firewall. – Rosamondrosamund 2/3, 2014 at 20:15

M

2

These incidents usually associated with NAT problems.

If you're sure that this isn't your problem, take a look at router configuration. Some routers are configured by default with "SIP ALG" option.

In some cases, this option should be off because implementation is incomplete.

Try it, and let me known if it works properly.

Muskmelon answered 25/8, 2014 at 9:45 Comment(2)

Thanks for your suggestion. Actually I figured it out earlier, It was due to NAT settings. I had resolved this issue. – Fussell 30/8, 2014 at 20:4

@Vivek How can you solved this problem with Nat ? What specific nat do you mean ?? – Luzluzader 6/7, 2021 at 14:51

S

0

make sure you have correct ip address in 'externip=' in sip.conf under /etc/asterisk.

Recommended topics

Hot tags