PLEASE HELP!- Asterisk hungs up after 7 seconds

BorjaGVO · December 6, 2011, 3:58pm

Hi, first of all I would like to say that I just started with Asterisk a month ago. I’m a completely newbie.
I’ve seen lot of people having this same issue. However, none of their solutions worked out for me. Yesterday I decided to reinstall Asterisk thinking that problem would go away (once, I had it working in some scenarios…).
Anyway, here is what I’m doing:

I have a NATed Asterisk and I’m doing tests everywhere. However, right now I’m testing it from home doing some calls from a SP (softphone-X-lite) inside LAN to another SP inside this same LAN (remember that Asterisk is in another LAN). Whenever a make a call it rings without a problem but there will be 2 problems:

The caller is not heared from the other side (while he does hear the other side)
The call lasts just a few seconds (7-10 sec.)

I went out over the sip trace setting debug on and I see two thing that make me suspect that something isn’t going well:
-At the beginning:

<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;received=192.168.1.1;rport=47388
From: sip:101@actualize.es;tag=92ad2c71
To: sip:103@actualize.es;tag=as04a28f41
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r344965M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="501b3342"
Content-Length: 0

At the end:

<------------->
— (9 headers 0 lines) —
Retransmitting #2 (NAT) to 192.168.1.1:47388:
BYE sip:101@192.168.1.1:47388 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport
Max-Forwards: 70
From: sip:103@actualize.es;tag=as09cec867
To: sip:101@actualize.es;tag=92ad2c71
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 103 BYE
User-Agent: Asterisk PBX SVN-branch-1.8-r344965M
Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:actualize.es”, nonce="", response="a9457c2d90bef8cf33e8333f970336d8"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0

My sip.conf (changed some options but the result is the same):

[b][general]
context=unauthenticated
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
localnet=192.168.1.0/255.255.255.0
externaddr=212.0.118.86
;directmedia=no

ofi
type=friend
context=LocalSets
nat=yes
reinvite=yes
qualify=yes
host=dynamic
;directmedia=nonat
secret=sip4ctu4l
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw

101
102
103
[/b]

Here it is the whole trace (101 calling 103) (Please feel free of asking me ANY question regarding my configuration):

[b]
<— SIP read from UDP:192.168.1.1:47388 —>
INVITE sip:103@actualize.es SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:101@192.168.1.1:47388
To: sip:103@actualize.es
From: sip:101@actualize.es;tag=92ad2c71
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63215
Content-Length: 378

v=0
o=- 1323185834370550 1 IN IP4 192.168.1.38
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.38
t=0 0
a=ice-ufrag:603986
a=ice-pwd:e9ef4090258b1b29a1aaf2d9e40fa869
m=audio 62146 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.38 62146 typ host
a=candidate:1 2 UDP 659134 192.168.1.38 62147 typ host
<------------->
— (13 headers 13 lines) —
Sending to 192.168.1.1:47388 (no NAT)
Using INVITE request as basis request - NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
Found peer ‘101’ for ‘101’ from 192.168.1.1:47388

<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;received=192.168.1.1;rport=47388
From: sip:101@actualize.es;tag=92ad2c71
To: sip:103@actualize.es;tag=as04a28f41
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r344965M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="501b3342"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.1:47388 —>
ACK sip:103@actualize.es SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;rport
Max-Forwards: 70
To: sip:103@actualize.es;tag=as04a28f41
From: sip:101@actualize.es;tag=92ad2c71
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.1.1:47388 —>
INVITE sip:103@actualize.es SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:101@192.168.1.1:47388
To: sip:103@actualize.es
From: sip:101@actualize.es;tag=92ad2c71
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63215
Authorization: Digest username=“101”,realm=“asterisk”,nonce=“501b3342”,uri="sip:103@actualize.es",response=“46208eb31f4b352ac1a16fd7c70880a0”,algorithm=MD5
Content-Length: 378

v=0
o=- 1323185834370550 1 IN IP4 192.168.1.38
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.38
t=0 0
a=ice-ufrag:603986
a=ice-pwd:e9ef4090258b1b29a1aaf2d9e40fa869
m=audio 62146 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.38 62146 typ host
a=candidate:1 2 UDP 659134 192.168.1.38 62147 typ host
<------------->
— (14 headers 13 lines) —
Sending to 192.168.1.1:47388 (NAT)
Using INVITE request as basis request - NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
Found peer ‘101’ for ‘101’ from 192.168.1.1:47388
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.38:62146
Looking for 103 in LocalSets (domain actualize.es)
list_route: hop: sip:101@192.168.1.1:47388

<— Transmitting (NAT) to 192.168.1.1:47388 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388
From: sip:101@actualize.es;tag=92ad2c71
To: sip:103@actualize.es
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r344965M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:103@192.168.1.199:5060
Content-Length: 0

<------------>
Audio is at 29466
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.1:36842:
INVITE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport
Max-Forwards: 70
From: “101” sip:101@192.168.1.199;tag=as65397079
To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c
Contact: sip:101@192.168.1.199:5060
Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r344965M
Date: Tue, 06 Dec 2011 15:37:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 224739662 224739662 IN IP4 192.168.1.199
s=Asterisk PBX SVN-branch-1.8-r344965M
c=IN IP4 192.168.1.199
t=0 0
m=audio 29466 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.1.1:36842 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport=5060;received=212.0.118.86
Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c
To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a
From: "101"sip:101@192.168.1.199;tag=as65397079
Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c

<— Transmitting (NAT) to 192.168.1.1:47388 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388
From: sip:101@actualize.es;tag=92ad2c71
To: sip:103@actualize.es;tag=as09cec867
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r344965M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:103@192.168.1.199:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.1.1:47388 —>

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

<— SIP read from UDP:192.168.1.1:36842 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport=5060;received=212.0.118.86
Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c
To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a
From: "101"sip:101@192.168.1.199;tag=as65397079
Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 377

v=0
o=- 12967659441998509 1 IN IP4 192.168.1.34
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.34
t=0 0
a=ice-ufrag:4ba8a2
a=ice-pwd:729e3ff03866436c713d52d18176b88d
m=audio 60798 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.34 60798 typ host
a=candidate:1 2 UDP 659134 192.168.1.34 60799 typ host
<------------->
— (12 headers 13 lines) —
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.34:60798
set_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to
set_destination: set destination to 192.168.1.1:36842
Transmitting (NAT) to 192.168.1.1:36842:
ACK sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0
Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK1724c7ae;rport
Max-Forwards: 70
From: “101” sip:101@192.168.1.199;tag=as65397079
To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a
Contact: sip:101@192.168.1.199:5060
Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.8-r344965M
Content-Length: 0

Audio is at 19120
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388
From: sip:101@actualize.es;tag=92ad2c71
To: sip:103@actualize.es;tag=as09cec867
Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.8-r344965M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:103@192.168.1.199:5060
Content-Type: application/sdp
Content-Length: 278