No response to our critical packet

Asterisk Asterisk Support
1

Hello
From the begining of our asterisk 1.4.41 installation we are suffering with an issue.
From time to time sip calls are hanging up at 20 seconds. I spent several hour in searching the reason in web but couldnt find a solution. There are several things written in forum but no change for me.

My asterisk server is not benhind firewall and Nat. this problem occured from time to time for connections that are behind and also for connections that are not behind nat

i found these in asterisk logs
Maximum retries exceeded on transmission NWNmOTk4N2FlZmE1YjU5NDdkM2M5MGNjMTFhNmU1NTY. for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
and the next sentence is

Hanging up call NWNmOTk4N2FlZmE1YjU5NDdkM2M5MGNjMTFhNmU1NTY. - no reply to our critical packet (see doc/sip-retransmit.txt).

Today i was on my pc and able to execute " sip set debug on " in asterisk cli and got the debug file

Looking forward to hearing a solution for this problem

Yours
Rock

here is the log

<— Reliably Transmitting (NAT) to 94.79.105.248:5062 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #1 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #2 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #3 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #4 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #5 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

Retransmitting #6 (NAT) to 94.79.105.248:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 94.79.105.248:5062;branch=z9hG4bK3d934bbd8d7590151;received=94.79.105.248

From: “95152” sip:95152@sip.polyphonecyprus.com:5060;tag=a81acafd40

To: sip:03924440111@sip.polyphonecyprus.com;tag=as09392fa6

Call-ID: 84a037ccb7e1168f

CSeq: 1906462937 INVITE

User-Agent: FPBX-2.7.0(1.4.41)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces

Contact: sip:03924440111@212.108.128.50

Content-Type: application/sdp

Content-Length: 286

v=0

o=root 26949 26950 IN IP4 212.108.128.50

s=session

c=IN IP4 212.108.128.50

t=0 0

m=audio 18118 RTP/AVP 0 8 96 125

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 iLBC/8000

a=fmtp:96 mode=30

a=rtpmap:125 telephone-event/8000

a=fmtp:125 0-16

a=ptime:20

a=sendrecv

2

94.79.105.248:5062 either failed to receive the INVITE packet, or failed to get its 100 Trying response back to Asterisk.

Why are you overriding the port number?

3

Dear david55

I didn’t configure anything to override The port number
Some sip clients connect from 5060 and some of them connect with a different port number

I really need help to solve this help

Yours
Rock

4

Did the port number come from a registration then?

If the port number is correct, the problem lies outside Asterisk.

5

Yes,
Port number and ip is correct and they come from the registration

This issue occurs from time to time . And when this happens all calls hang up at 20 seconds.

I use asterisk 1.4.41 for pure voip. Can it be related with the version of asterisk ?

Any help is appreciated

Yours
Rock

6

Extremely unlikely that it is Asterisk related. It is much more likey that a router has stopped forwarding in either the forward or reverse direction. I think it very unlikely that Asterisk would log writing a packet and not write it, and also very unlikely that it would fail to log receiving a packet on 5060, even if it was invalid.

The 20 seconds is just the timeout for when no response is received from the remote system when trying to initiate a call.

7

Dear David
What is your advice for me to find the solution.
What should i check
What I know that my Internet provider told me that my ip is not behind firewall and nat
I set nat=yes and canreinvite= no for all sip accounts and no chance everyday a few times I suffer from this problem.

Ossec is running on my server I don’t know whether this should be the reason

Yours
Huseyin

8

Take IP protocol traces at accessible points in the network to narrow down where the packet is getting lost and, if possible, whether inbound or outbound.

Make sure that you really understand the structure of the network; you should not need to ask the ISP about NAT and firewalls.

9

I’ve seen a similar issue when the firewall was closing udp session after 15 sec idle. You may want to allow the client fw to allow udp ‘sessions’ to live for 2 mins

10

Dear David55 and Cerien.Jean

I have little experience with asterisk.
What I have in hand is the log that I got from asterisk Cli with sip set debug on.
And at the same time I got sip traces at the client site.

The interesting point that this problem does occur from time to time
And when this happens it affects all the clients

I am looking for a professional to help me to solve this issue and ready to pay for the solution

Yours
Rock

11

It is nice to see someone willing to pay for help… too many people just expect you to do for free the job they’re paid for !

unfortunately, I am not available to support beyond this forum.

If the issue happens with all clients at the same time, then you have an issue with your link from asterisk to internet. It may go down, causing the calls to drop. If it is one client, it can be an issue with this client link, either dropping, or firewall / router issue.

You need to find more a network specialist than an asterisk specialist. You may want to contact whoever installed the network on the premises, or find locally an IT specialist with network skills.

KR
Jean

12

Did you ever get this figured out? I am having the same problem, but only with calls from one OpenSer servers out of the 2 in our system. Please let me know.

Thanks.