[SOLVED] Call drops after 6-7 seconds with no audio

Greetings,
I have just upgraded an Asterisk 1.6.2-9 to Asterisk 1.8.13-1 (Debian distribution) and started to notice a problem with some peers : calls drop after 6-7 seconds and I have no audio.
I have a lot of peers registered and experiencing the problem only with 2-3 of them. With Asterisk 1.6.2-9, same configuration everything was working fine.

Reliably Transmitting (NAT) to remotepeerip:13779:
OPTIONS sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK63260724;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@localasteriskpublicip:5061>;tag=as0babb18d
To: <sip:sippeeraccount@192.168.254.2:13779>
Contact: <sip:asterisk@localasteriskpublicip:5061>
Call-ID: 7bda9a255cfa742d561e2cf354618773@localasteriskpublicip:5061
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Wed, 23 Oct 2013 15:58:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
Supported: 100rel,timer
From: "asterisk"<sip:asterisk@localasteriskpublicip:5061>;tag=as0babb18d
To: <sip:sippeeraccount@192.168.254.2:13779>;tag=4C8E324631353641000ED474
Call-ID: 7bda9a255cfa742d561e2cf354618773@localasteriskpublicip:5061
CSeq: 102 OPTIONS
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK63260724;rport
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '7bda9a255cfa742d561e2cf354618773@localasteriskpublicip:5061' Method: OPTIONS

<--- SIP read from UDP:remotepeerip:13779 --->
INVITE sip:callednumber@localasteriskpublicip:5061 SIP/2.0
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DEB324631353641000ED497
To: <sip:callednumber@localasteriskpublicip:5061>
Contact: <sip:sippeeraccount@192.168.254.2:13779>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA1A33834ACA88F01
Content-Length: 302

v=0
o=- 0 0 IN IP4 192.168.95.15
s=T009
c=IN IP4 192.168.254.2
t=0 0
m=audio 19220 RTP/AVP 18 2 8 9 110
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->


--- (14 headers 16 lines) ---
Sending to remotepeerip:13779 (NAT)
Using INVITE request as basis request - 02047C97AE81400000000007@192.168.95.15
Found peer 'sippeeraccount' for 'sippeeraccount' from remotepeerip:13779

<--- Reliably Transmitting (NAT) to remotepeerip:13779 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA1A33834ACA88F01;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DEB324631353641000ED497
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as3cc07c13
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 1 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asteriskserver", nonce="71958b64"
Content-Length: 0


<------------>


Scheduling destruction of SIP dialog '02047C97AE81400000000007@192.168.95.15' in 8320 ms (Method: INVITE)


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 1 ACK
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DEB324631353641000ED497
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as3cc07c13
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA1A33834ACA88F01
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


<--- SIP read from UDP:remotepeerip:13779 --->
INVITE sip:callednumber@localasteriskpublicip:5061 SIP/2.0
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>
Contact: <sip:sippeeraccount@192.168.254.2:13779>
Content-Type: application/sdp
CSeq: 2 INVITE
Authorization: Digest username="sippeeraccount",realm="asteriskserver",algorithm=MD5,nonce="71958b64",opaque="",uri="sip:callednumber@localasteriskpublicip",response="0bedf404147721684439006289b26559"
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D
Content-Length: 302

v=0
o=- 0 0 IN IP4 192.168.95.15
s=T009
c=IN IP4 192.168.254.2
t=0 0
m=audio 19220 RTP/AVP 18 2 8 9 110
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->


--- (15 headers 16 lines) ---
Sending to remotepeerip:13779 (NAT)
Using INVITE request as basis request - 02047C97AE81400000000007@192.168.95.15
Found peer 'sippeeraccount' for 'sippeeraccount' from remotepeerip:13779
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 8


Found RTP audio format 9


Found RTP audio format 110
Found audio description format G729 for ID 18


Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 110
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)


Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.254.2:19220
Looking for callednumber in context-out (domain localasteriskpublicip)
list_route: hop: <sip:sippeeraccount@192.168.254.2:13779>


<--- Transmitting (NAT) to remotepeerip:13779 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Length: 0


<------------>


Audio is at 12522
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to remotepeerip:13779 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE


Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325679 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

<------------>


Audio is at 12522
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


<--- Reliably Transmitting (NAT) to remotepeerip:13779 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

<------------>


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


Retransmitting #1 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


Retransmitting #2 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


Retransmitting #3 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


Retransmitting #4 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


Retransmitting #5 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


[Oct 23 17:58:38] NOTICE[11740]: chan_sip.c:26326 sip_poke_peer: Still have a QUALIFY dialog active, deleting


Retransmitting #6 (NAT) to remotepeerip:13779:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKA3F8619DB1E4A39D;received=remotepeerip;rport=13779
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:callednumber@localasteriskpublicip:5061>
Content-Type: application/sdp
Content-Length: 272

v=0
o=root 1014325679 1014325680 IN IP4 localasteriskpublicip
s=CiscoCustom
c=IN IP4 localasteriskpublicip
t=0 0
m=audio 12522 RTP/AVP 18 8 110
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=sendrecv

---


<--- SIP read from UDP:remotepeerip:13779 --->
ACK sip:callednumber@localasteriskpublicip:5061 SIP/2.0
CSeq: 2 ACK
To: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
From: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
Max-Forwards: 70
User-Agent: NEC-i SL Series A3.01
Via: SIP/2.0/UDP 192.168.254.2:13779;branch=z9hG4bKE15D5D1FF47D10F8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


[Oct 23 17:58:40] WARNING[11740]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 02047C97AE81400000000007@192.168.95.15 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 8320ms with no response
[Oct 23 17:58:40] WARNING[11740]: chan_sip.c:3685 retrans_pkt: Hanging up call 02047C97AE81400000000007@192.168.95.15 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).


Scheduling destruction of SIP dialog '02047C97AE81400000000007@192.168.95.15' in 8320 ms (Method: INVITE)


set_destination: Parsing <sip:sippeeraccount@192.168.254.2:13779> for address/port to send to
set_destination: set destination to 192.168.254.2:13779


Reliably Transmitting (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---


Retransmitting #1 (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---


Retransmitting #2 (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---


Retransmitting #3 (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---


Retransmitting #4 (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---


Retransmitting #5 (NAT) to remotepeerip:13779:
BYE sip:sippeeraccount@192.168.254.2:13779 SIP/2.0
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Max-Forwards: 70
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
User-Agent: Asterisk
Proxy-Authorization: Digest username="sippeeraccount", realm="asteriskserver", algorithm=MD5, uri="sip:localasteriskpublicip", nonce="", response="7346a64853b82d3c0aa20d4c032d20f5"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---


<--- SIP read from UDP:remotepeerip:13779 --->
SIP/2.0 200 OK
From: <sip:callednumber@localasteriskpublicip:5061>;tag=as44706f18
To: "sippeeraccount"<sip:sippeeraccount@localasteriskpublicip>;tag=4DF9324631353641000ED498
Call-ID: 02047C97AE81400000000007@192.168.95.15
CSeq: 102 BYE
Server: NEC-i SL Series A3.01/2.1
Via: SIP/2.0/UDP localasteriskpublicip:5061;branch=z9hG4bK370ce006;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

I think it is some sort of NAT problem but can’t understand why in Asterisk 1.6 everything works fine and in Asterisk 1.8 not. The NAT on remote side is configured to accept any packets coming from asteriskpublicip and forward to the internal IP address : 192.168.254.2. Asterisk is configured on a public IP.
Thanks a lot!

The branch on the ACK doesn’t match that on the 200 OK. I think that is a protocal violation by the remote side. Disabling SIP pedantic mode (a general section option) may help.

Thanks a lot! You saved my day!