Help on Unable to create channel of type 'SIP' (cause 20 -

Asterisk Asterisk Support
1

Hi everyone ,
I have an Asterisk 1.8.10.1(~dfsg-1ubuntu1) instaled un Ubuntu Server 12.04 . , one Welltech Gateway Voip Wellgate 2540, 4FXO, 2FE and 2 FXS LinksysPAP2T .
Welltech FXS take one PSTN line for use in our Asterisk .
Config :
---------------sip.conf --------------------
[general]
context=default ; Default context for incoming calls
port=5060
bindaddr=0.0.0.0
maxexpirey=3600
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
nat=yes
allow=all
;------------telefon Dan Zup-----------
[10]
type=friend
secret=XXXXXXXXX
context=default
insecure=very
host=dynamic
nat=yes
username=10
dtmfmode=rfc2833
qualify=yes
canreinvite=no
;----------telefon Cristi CAE---------
[11]
type=friend
secret=XXXXXXXX
context=default
insecure=very
host=dynamic
nat=yes
username=11
dtmfmode=rfc2833
qualify=yes
;----------telefon Eugen-------------
[12]
type=friend
secret=XXXXXXXX
context=default
insecure=very
host=dynamic
nat=yes
username=12
qualify=yes
;---------VoIP GW GKoo-----
[031XXXXXXXX]
type=friend
secret=XXXXXXXX
context=default
insecure=very
host=dynamic
nat=yes
username=031XXXXXXXX
qualify=yes
canreinvite=no
-----extensions.conf----------------------
[general]
static=yes
writeprotect=no
[default]
;static=yes
;writeprotect=no
exten => 10,1,Dial(SIP/10)
exten => 10,n,Hangup()
exten => 11,1,Dial(SIP/11,tTr)
exten => 11,n,Hangup()
exten => 12,1,Dial(SIP/12,tTr)
exten => 12,n,Hangup()
exten => 13,1,Dial(SIP/13,tTr)
exten => 13,n,Hangup()
exten => 14,1,Dial(SIP/14,tTr)
exten => 14,n,Hangup()
exten => 100,1,Dial(SIP/100,tTr)
exten => 100,n,Hangup()
;VoIP GW Out
exten => _0XXXXXXXXX,1,Dial(SIP/${EXTEN}@0317305347,ftT)
exten => _0XXXXXXXXX,n,Hangup()
;VoIP GW In
;exten => 0317305347,1,Dial(SIP/100&SIP/10&SIP/11&SIP/13&SIP/14,80,ftT)
exten => 0317305347,1,Dial(SIP/10)
exten => 0317305347,n,Hangup()
-------------------------------------log---------------
Connected to Asterisk 1.8.10.1~dfsg-1ubuntu1 currently running on ms1 (pid = 27433)
Verbosity is at least 7
ms1*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
031XXXXXXXX/031XXXXXX 5.12.221.188 D N 8080 OK (70 ms)
10/10 80.86.114.70 D N 5060 OK (75 ms)
100/100 (Unspecified) D N 0 UNKNOWN
11/11 (Unspecified) D N 0 UNKNOWN
12/12 (Unspecified) D N 0 UNKNOWN
13/13 80.86.114.70 D N 1028 OK (75 ms)
14/14 (Unspecified) D N 0 UNKNOWN
7 sip peers [Monitored: 3 online, 4 offline Unmonitored: 0 online, 0 offline]
-------------------LOG debug sip--------------------------------

Adding codec 0x8000 (slin16) to SDP
Adding codec 0x200000000 (speex16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.86.114.70:5060:
INVITE sip:10@10.0.1.136:5060 SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK75c4dd96;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as06a0b5b6
To: sip:10@10.0.1.136:5060
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 03df852815a73f706297472f4a262a4e@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Thu, 18 Jul 2013 07:39:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 595

v=0
o=root 1797450780 1797450780 IN IP4 213.239.192.9
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 213.239.192.9
t=0 0
m=audio 13498 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/10

[Jul 18 09:39:20] WARNING[27118]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

<— SIP read from UDP:80.86.114.70:5060 —>
SIP/2.0 100 Trying
To: sip:10@10.0.1.136:5060
From: “anonymous” sip:anonymous@213.239.192.9;tag=as06a0b5b6
Call-ID: 03df852815a73f706297472f4a262a4e@213.239.192.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK75c4dd96
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0

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

<— SIP read from UDP:80.86.114.70:5060 —>
SIP/2.0 486 Busy Here
To: sip:10@10.0.1.136:5060;tag=5c734631c912bb3ei0
From: “anonymous” sip:anonymous@213.239.192.9;tag=as06a0b5b6
Call-ID: 03df852815a73f706297472f4a262a4e@213.239.192.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK75c4dd96
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0

<------------->
— (8 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 80.86.114.70:5060
Transmitting (NAT) to 80.86.114.70:5060:
ACK sip:10@10.0.1.136:5060 SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK75c4dd96;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as06a0b5b6
To: sip:10@10.0.1.136:5060;tag=5c734631c912bb3ei0
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 03df852815a73f706297472f4a262a4e@213.239.192.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0

-- SIP/10-00000047 is busy

== Everyone is busy/congested at this time (2:1/0/1)
– Executing [0317305347@default:2] Hangup(“SIP/0317305347-00000046”, “”) in new stack
== Spawn extension (default, 0317305347, 2) exited non-zero on 'SIP/0317305347-00000046’
Really destroying SIP dialog ‘03df852815a73f706297472f4a262a4e@213.239.192.9:5060’ Method: INVITE
Reliably Transmitting (NAT) to 80.86.114.70:5060:
OPTIONS sip:10@10.0.1.136:5060 SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK4f2f62f1;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@213.239.192.9;tag=as4d1c207c
To: sip:10@10.0.1.136:5060
Contact: sip:asterisk@213.239.192.9:5060
Call-ID: 63ece5e2764650e36cd115b0057938fd@213.239.192.9:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Thu, 18 Jul 2013 07:39:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:80.86.114.70:5060 —>
SIP/2.0 200 OK
To: sip:10@10.0.1.136:5060;tag=5c734631c912bb3ei0
From: “asterisk” sip:asterisk@213.239.192.9;tag=as4d1c207c
Call-ID: 63ece5e2764650e36cd115b0057938fd@213.239.192.9:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK4f2f62f1
Server: Linksys/PAP2-3.1.9(LSc)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘63ece5e2764650e36cd115b0057938fd@213.239.192.9:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 80.86.114.70:5060:
OPTIONS sip:10@10.0.1.136:5060 SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK5a50bf5f;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@213.239.192.9;tag=as58b43640
To: sip:10@10.0.1.136:5060
Contact: sip:asterisk@213.239.192.9:5060
Call-ID: 0d4217990f5cf8d45c963b9b57224f87@213.239.192.9:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Thu, 18 Jul 2013 07:40:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:80.86.114.70:5060 —>
SIP/2.0 200 OK
To: sip:10@10.0.1.136:5060;tag=5c734631c912bb3ei0
From: “asterisk” sip:asterisk@213.239.192.9;tag=as58b43640

…one week of trying , debugging and nothing found …
Can you be so kind and help ?
Asterisk have public ip ,both Linksys and Welltech behind nat .

2

I can dial out any number and work OK, but when dial my PSTN number from an mobilephone it rings and when i answer from extension 10 appear that messages and connection never made , which is correct because of 2 errors :
first : Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
second : Got SIP response 486 “Busy Here” back from 80.86.114.70:5060
What is wrong here ???

3

Busy here is an issue with the Linksys.

Cause 20 is because the alternative device is not registered.

Note that insecure=very does nothing with that version, which is probably a good thing for your security.

In general, you don’t seem to have read the security document!

4

[quote=“david55”]Busy here is an issue with the Linksys.

Cause 20 is because the alternative device is not registered.

Note that insecure=very does nothing with that version, which is probably a good thing for your security.

In general, you don’t seem to have read the security document![/quote]
Thank you very much . I try another test :
When i dial in and I have configured on a Samsung galaxy S3 Zoiper /SIP as extension 14 :
i try to answer and do not connect .
----------log----------------------------------------

   > Saved useragent "Zoiper r18976" for peer 14

<— Transmitting (NAT) to 80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.86.114.70:1030;branch=z9hG4bK-d8754z-0e23e9831a50fa64-1—d8754z-;received=80.86.114.70;rport=1030
From: sip:14@ms1.esserio.ro;transport=UDP;tag=2f572a3e
To: sip:14@ms1.esserio.ro;transport=UDP;tag=as31bae5d0
Call-ID: YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.
CSeq: 8 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;expires=60
Date: Fri, 19 Jul 2013 08:18:41 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK756d3629;rport=5060
Contact: sip:10.0.1.212:5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=63142774
From: "asterisk"sip:asterisk@213.239.192.9;tag=as03be568d
Call-ID: 5884ec1b0426489520d6b7e2385c1e71@213.239.192.9:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r18976
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘5884ec1b0426489520d6b7e2385c1e71@213.239.192.9:5060’ Method: OPTIONS

<— SIP read from UDP:80.86.114.70:1030 —>

Scheduling destruction of SIP dialog ‘335164266e8ac84a4a9cc78412277886@213.239.192.9:5060’ in 6400 ms (Method: INVITE)
Really destroying SIP dialog ‘0a343be516e2c54147cdc71454e12cfc@213.239.192.9:5060’ Method: INVITE
== Using SIP RTP CoS mark 5
– Executing [0317305347@default:1] Dial(“SIP/0317305347-00000008”, “SIP/100&SIP/10&SIP/11&SIP/13&SIP/14,80,ftT”) in new stack
[Jul 19 10:19:09] WARNING[18023]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
– Called SIP/10
[Jul 19 10:19:09] WARNING[18023]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
– Called SIP/13
== Using SIP RTP CoS mark 5
Audio is at 17712
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x200000000 (speex16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.86.114.70:1030:
INVITE sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Fri, 19 Jul 2013 08:19:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 595

v=0
o=root 1916374840 1916374840 IN IP4 213.239.192.9
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 213.239.192.9
t=0 0
m=audio 17712 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/14
-- SIP/13-0000000a is ringing
-- SIP/10-00000009 is ringing

Retransmitting #1 (NAT) to 80.86.114.70:1030:
INVITE sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Fri, 19 Jul 2013 08:19:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 595

v=0
o=root 1916374840 1916374840 IN IP4 213.239.192.9
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 213.239.192.9
t=0 0
m=audio 17712 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport=5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport=5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport=5060
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=c522ee3e
From: "anonymous"sip:anonymous@213.239.192.9;tag=as6786dfd0
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Zoiper r18976
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
– SIP/14-0000000b is ringing

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK3e3b35a2;rport=5060
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=c522ee3e
From: "anonymous"sip:anonymous@213.239.192.9;tag=as6786dfd0
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r18976
Allow-Events: presence, kpml
Content-Length: 259

v=0
o=Zoiper 0 2 IN IP4 80.86.114.70
s=Zoiper
c=IN IP4 80.86.114.70
t=0 0
m=audio 33000 RTP/AVP 0 110 3 8 98
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=sendrecv
<------------->
— (13 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 98
Found audio description format PCMU for ID 0
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 98
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x60e (gsm|ulaw|alaw|speex|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 80.86.114.70:33000
list_route: hop: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
set_destination: Parsing sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP for address/port to send to
set_destination: set destination to 80.86.114.70:1030
Transmitting (NAT) to 80.86.114.70:1030:
ACK sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK2ead171f;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=c522ee3e
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0

-- SIP/14-0000000b answered SIP/0317305347-00000008

Scheduling destruction of SIP dialog ‘190a43563ff1c040146d212211eb54dc@213.239.192.9:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP for address/port to send to
set_destination: set destination to 80.86.114.70:1030
Reliably Transmitting (NAT) to 80.86.114.70:1030:
BYE sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK285a023d;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as6786dfd0
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=c522ee3e
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (default, 0317305347, 1) exited non-zero on ‘SIP/0317305347-00000008’

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK285a023d;rport=5060
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=c522ee3e
From: "anonymous"sip:anonymous@213.239.192.9;tag=as6786dfd0
Call-ID: 190a43563ff1c040146d212211eb54dc@213.239.192.9:5060
CSeq: 103 BYE
User-Agent: Zoiper r18976
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘190a43563ff1c040146d212211eb54dc@213.239.192.9:5060’ Method: INVITE
Really destroying SIP dialog ‘YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.’ Method: REGISTER
Really destroying SIP dialog ‘335164266e8ac84a4a9cc78412277886@213.239.192.9:5060’ Method: INVITE

<— SIP read from UDP:80.86.114.70:1030 —>

<------------->
ms1*CLI>

m=audio 15830 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport=5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
From: “anonymous” sip:anonymous@213.239.192.9;tag=as78e44d9d
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport=5060
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=765b4778
From: "anonymous"sip:anonymous@213.239.192.9;tag=as78e44d9d
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Zoiper r18976
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: hop: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
– SIP/14-00000008 is ringing
Scheduling destruction of SIP dialog ‘05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP for address/port to send to
set_destination: set destination to 80.86.114.70:1030
Reliably Transmitting (NAT) to 80.86.114.70:1030:
CANCEL sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as78e44d9d
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0

Scheduling destruction of SIP dialog ‘05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060’ in 6400 ms (Method: INVITE)
== Spawn extension (default, 0317305347, 1) exited non-zero on 'SIP/0317305347-00000006’
Retransmitting #1 (NAT) to 80.86.114.70:1030:
CANCEL sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as78e44d9d
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport=5060
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=765b4778
From: "anonymous"sip:anonymous@213.239.192.9;tag=as78e44d9d
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 CANCEL
User-Agent: Zoiper r18976
Content-Length: 0

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

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport=5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=765b4778
From: "anonymous"sip:anonymous@213.239.192.9;tag=as78e44d9d
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 INVITE
User-Agent: Zoiper r18976
Content-Length: 0

<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:14@80.86.114.70:1030 for address/port to send to
set_destination: set destination to 80.86.114.70:1030
Transmitting (NAT) to 80.86.114.70:1030:
ACK sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@213.239.192.9;tag=as78e44d9d
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=765b4778
Contact: sip:anonymous@213.239.192.9:5060
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0

Scheduling destruction of SIP dialog ‘05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK77b61378;rport=5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=e35a024e
From: "anonymous"sip:anonymous@213.239.192.9;tag=as78e44d9d
Call-ID: 05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060
CSeq: 102 CANCEL
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘05c419e328dc502f23e92fa3392d2b39@213.239.192.9:5060’ Method: INVITE

<— SIP read from UDP:80.86.114.70:1030 —>
REGISTER sip:ms1.esserio.ro;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 80.86.114.70:1030;branch=z9hG4bK-d8754z-1dc37aded14d7334-1—d8754z-
Max-Forwards: 70
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@ms1.esserio.ro;transport=UDP
From: sip:14@ms1.esserio.ro;transport=UDP;tag=2f572a3e
Call-ID: YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.
CSeq: 42 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r18976
Authorization: Digest username=“14”,realm=“asterisk”,nonce=“6b227c07”,uri=“sip:ms1.esserio.ro;transport=UDP”,response=“d5f138770bcaabb7e62d5013bd8016ee”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 80.86.114.70:1030 (NAT)

<— Transmitting (NAT) to 80.86.114.70:1030 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 80.86.114.70:1030;branch=z9hG4bK-d8754z-1dc37aded14d7334-1—d8754z-;received=80.86.114.70;rport=1030
From: sip:14@ms1.esserio.ro;transport=UDP;tag=2f572a3e
To: sip:14@ms1.esserio.ro;transport=UDP;tag=as1954ed80
Call-ID: YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.
CSeq: 42 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0a14d172"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:80.86.114.70:1030 —>
REGISTER sip:ms1.esserio.ro;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 80.86.114.70:1030;branch=z9hG4bK-d8754z-f5fa79ae43d0b043-1—d8754z-
Max-Forwards: 70
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
To: sip:14@ms1.esserio.ro;transport=UDP
From: sip:14@ms1.esserio.ro;transport=UDP;tag=2f572a3e
Call-ID: YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.
CSeq: 43 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r18976
Authorization: Digest username=“14”,realm=“asterisk”,nonce=“0a14d172”,uri=“sip:ms1.esserio.ro;transport=UDP”,response=“c8038715041f7e385df0a8f9423ee2a3”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 80.86.114.70:1030 (NAT)
Reliably Transmitting (NAT) to 80.86.114.70:1030:
OPTIONS sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK17579546;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@213.239.192.9;tag=as1a79ea70
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP
Contact: sip:asterisk@213.239.192.9:5060
Call-ID: 4cc337403b30c7d33c382b805682a242@213.239.192.9:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Fri, 19 Jul 2013 08:36:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to 80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.86.114.70:1030;branch=z9hG4bK-d8754z-f5fa79ae43d0b043-1—d8754z-;received=80.86.114.70;rport=1030
From: sip:14@ms1.esserio.ro;transport=UDP;tag=2f572a3e
To: sip:14@ms1.esserio.ro;transport=UDP;tag=as1954ed80
Call-ID: YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.
CSeq: 43 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;expires=60
Date: Fri, 19 Jul 2013 08:36:34 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YTkyMDZhZmYzZWQ5YWYxZTc2N2Y2YzJkYTdiYTIzNDY.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:80.86.114.70:1030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.239.192.9:5060;branch=z9hG4bK17579546;rport=5060
Contact: sip:10.0.1.212:5060
To: sip:14@80.86.114.70:1030;rinstance=f4d33daf31dc8de4;transport=UDP;tag=f939fe5e
From: "asterisk"sip:asterisk@213.239.192.9;tag=as1a79ea70
Call-ID: 4cc337403b30c7d33c382b805682a242@213.239.192.9:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Zoiper r18976
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘4cc337403b30c7d33c382b805682a242@213.239.192.9:5060’ Method: OPTIONS

<— SIP read from UDP:80.86.114.70:1030 —>

<------------->
ms1*CLI>

So everithing I try to dial in do not work : is there some curse …!!!