The usual suspects, Call drop with SIP retransmission, but no NAT

Asterisk Asterisk Support
1

Hi everyone,
I’ve always thought I have at least mastered the basics of asterisk but then a problem like this teaches me that I’m still a noob after all… Enough of wailing, maybe, you can help me with this problem

Customer wants to connect a Siedle Access Server doorphone using Asterisk.
I have palced a RasPi using with Asterisk at the customers Site. The RasPi has an IAX Link to the provider.
As RasPi and Doorphone are in the same subnet there is no NAT involved whatsoever.

When I test the setup with a Yealink phone everything works like a charm.
When I register the Siedle Box, I get two-way audio, but then the call drops after 6 secs or 32secs (depending on the qualify parameter in sip.conf)…

Almost all information that I could find on the net is connected with NAT-related problems.
Can anyone of you guys see what’s wrong based on the data below? Any help will be greatly appreciated.
Btw. I will not paste the whole SIP dialog, just not clog up the forums, but if you need it for the whole Picture, let me know…

The console message is the infamous:
[Mar 31 06:45:13] WARNING[491]: chan_sip.c:4028 retrans_pkt: Retransmission timeout reached on transmission 98946a0d-c0e8-4a47-99f6-0c2ffd0cd1b1@192.168.140.253 for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Mar 31 06:45:13] WARNING[491]: chan_sip.c:4057 retrans_pkt: Hanging up call 98946a0d-c0e8-4a47-99f6-0c2ffd0cd1b1@192.168.140.253 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– Hungup ‘IAX2/Front2Back88-8908’

  1. Asterisk: Asterisk 11.13.1~dfsg-2+b1
  2. sip.conf (the rest of the default sip.conf has been unchanged - I guess its pretty much minimum requirements what I’ve configured here)

[0335222272110]
type=friend
qualify=yes
language=de
host=dynamic
user=0335222272110
secret=notofrelevanceforthisproblem
context=ausgehend
callerid=“Tuersprechstelle” <10>
canreinvite=no
dtmfmode=rfc2833
allow=alaw,gsm,ulaw
language=de
qualify=yes
directrtpsetup=no

Retransmitted packets
Those are retransmitted 6 times before the call drops…
The communication before seems pretty standard - if you need it, I will be happy to provide the data…

Retransmitting #2 (no NAT) to 192.168.140.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.253:5060;branch=z9hG4bK-4190121481;received=192.168.140.253;rport=5060
From: sip:031626972110@192.168.140.252;tag=4190121502
To: sip:00436642029902@192.168.140.252;tag=as72b57329
Call-ID: 5430514d-7999-394d-9544-a27aeb3b8845@192.168.140.253
CSeq: 2 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:00436642029902@192.168.140.252:5060
Content-Type: application/sdp
Content-Length: 321

v=0
o=root 1845817189 1845817189 IN IP4 192.168.140.252
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 192.168.140.252
t=0 0
m=audio 13906 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99

eRetransmitting #3 (no NAT) to 192.168.140.253:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.140.253:5060;branch=z9hG4bK-4190121481;received=192.168.140.253;rport=5060
From: sip:031626972110@192.168.140.252;tag=4190121502
To: sip:00436642029902@192.168.140.252;tag=as72b57329
Call-ID: 5430514d-7999-394d-9544-a27aeb3b8845@192.168.140.253
CSeq: 2 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:00436642029902@192.168.140.252:5060
Content-Type: application/sdp
Content-Length: 321

v=0
o=root 1845817189 1845817189 IN IP4 192.168.140.252
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 192.168.140.252
t=0 0
m=audio 13906 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99

Greetings from Austria
George

2

Please use preformatted text, otherwise important information, including the contents of the Contact header, is lost.

Asterisk is not receiving an ACK to its response. Is 192.168.140.253:5060 the correct address for the device? Is the contents of the Contact header the correct address for reaching Asterisk from the device?

You have the normal selection of deprecated and obsolete parameters in your sip.conf and inappropriate use of friend, which typically result from using outdated, and probably always wrong, cookbook configurations. However, none are relevant here.

3

Hi Everyone

First of all: David, thank you for you answer and sorry for the text mess, I need to get used to the new forums :wink:
I can confirm that the doorphone has the specified IP-Adress
I have - of course been trying around a lot and inserting from ‘cookbook configs’ that you have mentioned what seemed to be loosely connected to my problem ;-).
BTW: As The doorphone should also be able to accept calls (please do not ask why), I went for friend.

My original sip.conf was

[031626972110]
type=friend
host=dynamic
qualify=yes
language=de
secret=notrelevanthere
context=ausgehend
callerid="Tuersprechstelle" <10>
canreinvite=no
dtmfmode=rfc2833

In the meanwhile I have found out:
I am rendering the packet flow here without a dump, so that it is easier
Packets:
Doorphone/Asterisk ->INVITE-> <-401 Unauthorized<- ->INVITE-> <-100 TRYING<- <-100 TRYING<- <-180 RINGING<- <-183 Session Progress<- <-200 OK with session Description<- ->ACK-> <-200 OK with session Description<- <-200 OK with session Description<- <-200 OK with session Description<- <-200 OK with session Description<- <-200 OK with session Description<- <-BYE<- ->OK->

I hope you do not mind me posting all of the dialog here, but I think it will maybe enable you to see what I am missing.
My two wild guesses are

4

SOLVED!

And the Kudos go to David55 (thanks again, mate you lead me in the right track) for his Post in the old forums: http://forums.asterisk.org/viewtopic.php?f=1&t=88494

It turns out that insecure=invite solves the problem
My sip.conf section currently looks like this:

[031626972110]
type=friend
qualify=yes
language=de
host=dynamic
secret=hasbeenchanged
context=ausgehend
callerid="Tuersprechstelle " <10>
canreinvite=no
dtmfmode=rfc2833
insecure=invite