Asterisk and Skype Connect Issue

divito · April 9, 2014, 9:25am

Hi everyone, I am trying to communicate between asterisk and skype connect. I have a sip profile registered with 1 channel. also, asterisk registers to sip.skype.com ok. From the CLI:

sip show peers

skype/99051000XXXXXX 63.209.144.201 Yes Yes 5060 OK (178 ms)

sip show registry

Reg.Time sip.skype.com:5060 N 99051000XXXX 30 Registered Tue, 08 Apr 2014 22:18:29 1 SIP registrations.

However, whenever I try to make a call to skype, I get:

*CLI> == Using SIP RTP CoS mark 5 -- Executing [4321@LocalSets:1] Dial("SIP/ZOI6001-00000038", "SIP/skype/4321") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/skype/4321 == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/ZOI6001-00000038' status is 'CHANUNAVAIL'

For some reason I think I should mention athat whenever i reload sip, it says Using Cos mark 4 and not Cos mark 5 like above.

In sip.conf:

[code][general]
;register => gnext.telephony:56w2t5EAeCfnum@sip.skype.com/99051000234871
context=from-trunk
allowoverlap=no
;allowguest=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
allow=all
dtmfmode=rfc2833
;bindport = 56782
port=5060
;register => gnext.telephony:56w2t5EAeCfnum@sip.skype.com/99051000234871
register =>99051000234871:56w2t5EAeCfnum@sip.skype.com/99051000234871
;register => 9905100xxxxxxx:xxxxxxxxxx@sip.skype.com/9905100xxxxxxxxx
trustrpid=no
sendrpid=yes
calllimit=4
defaultexpiry=240

[skype]
type=friend
;type=peer
;context=from-trunk
context=from-trunk
username=99051000234871
secret=56w2t5EAeCfnum
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
host=sip.skype.com
nat=force_rport,comedia
qualify=yes
fromuser=xxxxxxxxxx
fromdomain=sip.skype.com
disallow=all
allow=g729
allow=ulaw
allow=alaw
[/code]

My dialplan:

[from-trunk]
exten => 1234,n,Dial(SIP/XLX6003,15);

[from-local]
exten => 4321,1,Dial(SIP/skype/${EXTEN})

Sip debug (sip set debug on) shows:

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '4e3bd3734b8db5a35bf963ee50b0e2b3@192.168.0.103:5060' Method: OPTIONS
[Apr  8 22:26:18] NOTICE[12076]: chan_sip.c:15059 sip_reregister:    -- Re-registration for  99051000234871@sip.skype.com
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 63.209.144.201:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK06df1bdd
Max-Forwards: 70
From: <sip:99051000234871@sip.skype.com>;tag=as22c0d1e3
To: <sip:99051000234871@sip.skype.com>
Call-ID: 53a73ba5083511cb29b1d3cf6bf4f37b@[::1]
CSeq: 109 REGISTER
User-Agent: Asterisk PBX 11.8.1
Expires: 120
Contact: <sip:99051000234871@192.168.0.103:5060>
Content-Length: 0

any ideas? Please let me know.
Thanks!!

dejanst · April 9, 2014, 10:11am

It looks like Skype is rejecting your calls for whatever reason.

Please do a “sip set debug on”, make an outgoing call and copy/paste the Asterisk CLI output. You current debug does not show a call.

divito · April 9, 2014, 1:46pm

Ok dejanst,

So here is the debug information. seems the call goes into time-out, not sure why. Seems there’s and issue with unsupported events. I am making the calls from a zoiper softphone so ill check that out in the meantime.

*CLI> sip set debug on
SIP Debugging enabled
*CLI>
<--- SIP read from UDP:192.168.0.100:58657 --->


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

<--- SIP read from UDP:192.168.0.100:55916 --->


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

<--- SIP read from UDP:192.168.0.100:58657 --->
PUBLISH sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-50594bdca910a4cc-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=ff79d446
Call-ID: OGYyZWNkMmE4NzkwMzljZTVmMzZiMjJhZWViMjQ4YjI.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence
Allow-Events: presence, kpml
Content-Length: 274

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:ZOI6001@192.168.01.103                                                                                        ;transport=UDP"> <tuple id="ZOI6001" > <status><basic>open</basic></status> <not                                                                                        e>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.100:58657 (NAT)

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-50594bdca910a4cc-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=ff79d446
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as529058f2
Call-ID: OGYyZWNkMmE4NzkwMzljZTVmMzZiMjJhZWViMjQ4YjI.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'OGYyZWNkMmE4NzkwMzljZTVmMzZiMjJhZWViMjQ4YjI.' Meth                                                                                        od: PUBLISH

<--- SIP read from UDP:192.168.0.100:58657 --->
SUBSCRIBE sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-484159019b631ff6-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=a859ba64
Call-ID: MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYmI.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Creating new subscription
Sending to 192.168.0.100:58657 (NAT)
list_route: hop: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-484159019b631ff6-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=a859ba64
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as1a47949b
Call-ID: MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYmI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="397623a0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYm                                                                                        I.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.0.100:58657 --->
SUBSCRIBE sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-091c5cd2a5aeb3ce-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=a859ba64
Call-ID: MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYmI.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Authorization: Digest username="ZOI6001",realm="asterisk",nonce="397623a0",uri="                                                                                        sip:ZOI6001@192.168.01.103;transport=UDP",response="af15ea16fecc57bd38eccb9ff778                                                                                        5793",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.0.100:58657 (NAT)
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-091c5cd2a5aeb3ce-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=a859ba64
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as1a47949b
Call-ID: MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYmI.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'MGViYmQ0NDI0NDcwZTI0M2VhNWQ0MWRiMTdhOTZlYmI.' Meth                                                                                        od: SUBSCRIBE

<--- SIP read from UDP:192.168.0.100:58657 --->
INVITE sip:4321@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-7be2907a20716e97-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: <sip:4321@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 234

v=0
o=Z 0 0 IN IP4 65.183.8.50
s=Z
c=IN IP4 65.183.8.50
t=0 0
m=audio 26412 RTP/AVP 98 110 8 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Sending to 192.168.0.100:58657 (NAT)
Using INVITE request as basis request - MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJh                                                                                        ZDI.
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Reliably Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-7be2907a20716e97-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
To: <sip:4321@192.168.01.103;transport=UDP>;tag=as785932f4
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d5e9c25"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZD                                                                                        I.' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.100:58657 --->
ACK sip:4321@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-7be2907a20716e97-1---d8                                                                                        754z-;rport
Max-Forwards: 70
To: <sip:4321@192.168.01.103;transport=UDP>;tag=as785932f4
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.100:58657 --->
INVITE sip:4321@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-24c9228e05a4c793-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: <sip:4321@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Authorization: Digest username="ZOI6001",realm="asterisk",nonce="4d5e9c25",uri="                                                                                        sip:4321@192.168.01.103;transport=UDP",response="a7ca9f6e43421fecbb670662cc04b86                                                                                        f",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 234

v=0
o=Z 0 0 IN IP4 65.183.8.50
s=Z
c=IN IP4 65.183.8.50
t=0 0
m=audio 26412 RTP/AVP 98 110 8 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Using INVITE request as basis request - MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJh                                                                                        ZDI.
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657
  == Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 101
Found audio description format iLBC for ID 98
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw|speex|ilbc)/video=(nothi                                                                                        ng)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon                                                                                        e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 65.183.8.50:26412
Looking for 4321 in LocalSets (domain 192.168.01.103)
list_route: hop: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-24c9228e05a4c793-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
To: <sip:4321@192.168.01.103;transport=UDP>
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:4321@192.168.0.103:5060>
Content-Length: 0


<------------>
    -- Executing [4321@LocalSets:1] Dial("SIP/ZOI6001-00000004", "SIP/skype/4321                                                                                        ") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11374
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5060:
INVITE sip:4321@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK0d043207;rport
Max-Forwards: 70
From: "Divito" <sip:Divito.martin.ba@63.209.144.201>;tag=as7e696068
To: <sip:4321@sip.skype.com>
Contact: <sip:Divito.martin.ba@192.168.0.103:5060>
Call-ID: 0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Wed, 09 Apr 2014 15:43:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Remote-Party-ID: "Divito" <sip:ZOI6001@63.209.144.201>;party=calling;privacy=off                                                                                        ;screen=no
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 513359814 513359814 IN IP4 192.168.0.103
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.103
t=0 0
m=audio 11374 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/skype/4321

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 100 Trying
From: "Divito" <sip:Divito.martin.ba@63.209.144.201>;tag=as7e696068
To: <sip:4321@sip.skype.com>
Call-ID: 0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK0d043207;rport=27000;received=                                                                                        65.183.8.50
Contact: <sip:4321@sip.skype.com:5060;maddr=63.209.144.201;transport=udp>
Content-Length: 0

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

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 404 Not Found
From: "Divito" <sip:Divito.martin.ba@63.209.144.201>;tag=as7e696068
To: <sip:4321@sip.skype.com>;tag=ca90d13f-13c4-53454e95-d71d6840-b7b491a
Call-ID: 0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK0d043207;rport=27000;received=                                                                                        65.183.8.50
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 63.209.144.201:5060:
ACK sip:4321@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK0d043207;rport
Max-Forwards: 70
From: "Divito" <sip:Divito.martin.ba@63.209.144.201>;tag=as7e696068
To: <sip:4321@sip.skype.com>;tag=ca90d13f-13c4-53454e95-d71d6840-b7b491a
Contact: <sip:Divito.martin.ba@192.168.0.103:5060>
Call-ID: 0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '0b58a0486f5bb3f72b4f55f97546333c@63.209.14                                                                                        4.201' in 10176 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/ZOI6001-00000004' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-24c9228e05a4c793-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
To: <sip:4321@192.168.01.103;transport=UDP>;tag=as5ff3a087
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.100:58657 --->
ACK sip:4321@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-24c9228e05a4c793-1---d8                                                                                        754z-;rport
Max-Forwards: 70
To: <sip:4321@192.168.01.103;transport=UDP>;tag=as5ff3a087
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=5863e65a
Call-ID: MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'MDQ5OGYxNjA4ZTAxZTliNDBjM2U0ZWMxNDNkMjJhZDI.' Meth                                                                                        od: ACK

<--- SIP read from UDP:192.168.0.100:58657 --->
PUBLISH sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-29f7e0e70e43a64e-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=f405822b
Call-ID: MDY3MDM4MzllOTRjYTQ5ZTkwM2FhMWRkMzIxMzhiMjI.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence
Allow-Events: presence, kpml
Content-Length: 268

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:ZOI6001@192.168.01.103                                                                                        ;transport=UDP"> <tuple id="ZOI6001" > <status><basic>open</basic></status> <not                                                                                        e>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 192.168.0.100:58657 (NAT)

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-29f7e0e70e43a64e-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=f405822b
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as2cdcfe5c
Call-ID: MDY3MDM4MzllOTRjYTQ5ZTkwM2FhMWRkMzIxMzhiMjI.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'MDY3MDM4MzllOTRjYTQ5ZTkwM2FhMWRkMzIxMzhiMjI.' Meth                                                                                        od: PUBLISH

<--- SIP read from UDP:192.168.0.100:58657 --->
SUBSCRIBE sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-d502fd254c6f3d25-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=da30560f
Call-ID: ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5YjY.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Creating new subscription
Sending to 192.168.0.100:58657 (NAT)
list_route: hop: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-d502fd254c6f3d25-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=da30560f
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as4033d3da
Call-ID: ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5YjY.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ec38cad"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5Yj                                                                                        Y.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.0.100:58657 --->
SUBSCRIBE sip:ZOI6001@192.168.01.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-041a3d84a1491024-1---d8                                                                                        754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=da30560f
Call-ID: ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5YjY.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Authorization: Digest username="ZOI6001",realm="asterisk",nonce="2ec38cad",uri="                                                                                        sip:ZOI6001@192.168.01.103;transport=UDP",response="a8f8b9f8130ac061987dc326224c                                                                                        e382",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 192.168.0.100:58657 (NAT)
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-041a3d84a1491024-1---d8                                                                                        754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=da30560f
To: "Divito"<sip:ZOI6001@192.168.01.103;transport=UDP>;tag=as4033d3da
Call-ID: ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5YjY.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS                                                                                        H
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'ZTdlMjY2ZDY3ZDU1OTgzZDU3OWM2ZDU5OGYwYWY5YjY.' Meth                                                                                        od: SUBSCRIBE
Really destroying SIP dialog '0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201' Method: INVITE
[Apr  9 10:43:46] NOTICE[2288]: chan_sip.c:15059 sip_reregister:    -- Re-registration for  99051000234871@sip.skype.com
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 63.209.144.201:5060:
REGISTER sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK45ceee8f
Max-Forwards: 70
From: <sip:99051000234871@sip.skype.com>;tag=as5b9f11b0
To: <sip:99051000234871@sip.skype.com>
Call-ID: 35075a5e6a43e3a109f4cbeb6d0db7f9@[::1]
CSeq: 108 REGISTER
User-Agent: Asterisk PBX 11.8.1
Expires: 120
Contact: <sip:99051000234871@192.168.0.103:5060>
Content-Length: 0


---

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 200 OK
From: <sip:99051000234871@sip.skype.com>;tag=as5b9f11b0
To: <sip:99051000234871@sip.skype.com>;tag=c990d13f-33536b5-0-46bb20c7-0
Call-ID: 35075a5e6a43e3a109f4cbeb6d0db7f9@[::1]
CSeq: 108 REGISTER
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK45ceee8f;rport=27000;received=65.183.8.50
Expires: 45
Contact: <sip:99051000234871@192.168.0.103:5060>;expires=45
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
[Apr  9 10:43:46] NOTICE[2288]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
Really destroying SIP dialog '35075a5e6a43e3a109f4cbeb6d0db7f9@[::1]' Method: REGISTER

abw1oim · April 10, 2014, 3:35pm

The packet.

[quote]<— SIP read from UDP:63.209.144.201:5060 —>
SIP/2.0 404 Not Found
From: “Divito” sip:Divito.martin.ba@63.209.144.201;tag=as7e696068
To: sip:4321@sip.skype.com;tag=ca90d13f-13c4-53454e95-d71d6840-b7b491a
Call-ID: 0b58a0486f5bb3f72b4f55f97546333c@63.209.144.201
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK0d043207;rport=27000;received= 65.183.8.50
Content-Length: 0[/quote]

shows Your problem: The destination 4321, You’re trying to reach at sip.skype.com is unknown there.
So make sure, the destination is existing there, otherwise it won’t work.
The “SIP/2.0 404 Not Found” is usually only thrown in cases where the destination dialed is not reachable through the specified trunk.

divito · April 10, 2014, 9:25pm

Abw1oim,

I think that is because I have ${EXTEN} in:

[from-local]
exten => 4321,1,Dial(SIP/skype/${EXTEN})

so when I dial 4321 from the softphone, it goes to:

exten => 4321,1,Dial(SIP/skype/4321)

However, when I set it to,
exten => 4321,1,Dial(SIP/skype/myskypeaccount)

I get errors all the same

*CLI> sip set debug on
SIP Debugging enabled
*CLI>
<--- SIP read from UDP:192.168.0.100:58657 --->


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

<--- SIP read from UDP:192.168.0.100:58657 --->
INVITE sip:4321@192.168.0.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-87610e44ff16c689-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: <sip:4321@192.168.0.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 1 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: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 234

v=0
o=Z 0 0 IN IP4 65.183.8.50
s=Z
c=IN IP4 65.183.8.50
t=0 0
m=audio 26412 RTP/AVP 98 110 8 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Sending to 192.168.0.100:58657 (NAT)
Using INVITE request as basis request - NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657

<--- Reliably Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-87610e44ff16c689-1---d8754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
To: <sip:4321@192.168.0.103;transport=UDP>;tag=as2b78d640
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 1 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f12b78a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.100:58657 --->
ACK sip:4321@192.168.0.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-87610e44ff16c689-1---d8754z-;rport
Max-Forwards: 70
To: <sip:4321@192.168.0.103;transport=UDP>;tag=as2b78d640
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.100:58657 --->
INVITE sip:4321@192.168.0.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-11d79e29a0357f32-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>
To: <sip:4321@192.168.0.103;transport=UDP>
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 2 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: Z 3.2.21357 r21367
Authorization: Digest username="ZOI6001",realm="asterisk",nonce="2f12b78a",uri="sip:4321@192.168.0.103;transport=UDP",response="86db080f2c5ddd7dddbba633c77139ce",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 234

v=0
o=Z 0 0 IN IP4 65.183.8.50
s=Z
c=IN IP4 65.183.8.50
t=0 0
m=audio 26412 RTP/AVP 98 110 8 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.0.100:58657 (NAT)
Using INVITE request as basis request - NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
Found peer 'ZOI6001' for 'ZOI6001' from 192.168.0.100:58657
  == Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 101
Found audio description format iLBC for ID 98
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g722), peer - audio=(alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 65.183.8.50:26412
Looking for 4321 in LocalSets (domain 192.168.0.103)
list_route: hop: <sip:ZOI6001@65.183.8.50:26417;transport=UDP>

<--- Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-11d79e29a0357f32-1---d8754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
To: <sip:4321@192.168.0.103;transport=UDP>
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:4321@192.168.0.103:5060>
Content-Length: 0


<------------>
    -- Executing [4321@LocalSets:1] Dial("SIP/ZOI6001-00000002", "SIP/skype/navito.martin") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11390
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5060:
INVITE sip:navito.martin@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK264d7a97;rport
Max-Forwards: 70
From: "Divito" <sip:navito.martin@63.209.144.201>;tag=as4879fc5b
To: <sip:navito.martin@sip.skype.com>
Contact: <sip:navito.martin@192.168.0.103:5060>
Call-ID: 2dd6f23537eebf0409997eb267677633@63.209.144.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.8.1
Date: Thu, 10 Apr 2014 23:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Divito" <sip:ZOI6001@63.209.144.201>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 308

v=0
o=root 2116565194 2116565194 IN IP4 192.168.0.103
s=Asterisk PBX 11.8.1
c=IN IP4 192.168.0.103
t=0 0
m=audio 11390 RTP/AVP 8 18 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/skype/navito.martin

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 100 Trying
From: "Divito" <sip:navito.martin@63.209.144.201>;tag=as4879fc5b
To: <sip:navito.martin@sip.skype.com>
Call-ID: 2dd6f23537eebf0409997eb267677633@63.209.144.201
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK264d7a97;rport=27000;received=65.183.8.50
Contact: <sip:navito.martin@sip.skype.com:5060;maddr=63.209.144.201;transport=udp>
Content-Length: 0

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

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 404 Not Found
From: "Divito" <sip:navito.martin@63.209.144.201>;tag=as4879fc5b
To: <sip:navito.martin@sip.skype.com>;tag=ca90d13f-13c4-53470b44-dde649a1-61fbfd03
Call-ID: 2dd6f23537eebf0409997eb267677633@63.209.144.201
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK264d7a97;rport=27000;received=65.183.8.50
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 63.209.144.201:5060:
ACK sip:navito.martin@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK264d7a97;rport
Max-Forwards: 70
From: "Divito" <sip:navito.martin@63.209.144.201>;tag=as4879fc5b
To: <sip:navito.martin@sip.skype.com>;tag=ca90d13f-13c4-53470b44-dde649a1-61fbfd03
Contact: <sip:navito.martin@192.168.0.103:5060>
Call-ID: 2dd6f23537eebf0409997eb267677633@63.209.144.201
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0


---
Scheduling destruction of SIP dialog '2dd6f23537eebf0409997eb267677633@63.209.144.201' in 20672 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/ZOI6001-00000002' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.0.100:58657 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-11d79e29a0357f32-1---d8754z-;received=192.168.0.100;rport=58657
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
To: <sip:4321@192.168.0.103;transport=UDP>;tag=as4781bb20
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.100:58657 --->
ACK sip:4321@192.168.0.103;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-11d79e29a0357f32-1---d8754z-;rport
Max-Forwards: 70
To: <sip:4321@192.168.0.103;transport=UDP>;tag=as4781bb20
From: "Divito"<sip:ZOI6001@192.168.0.103;transport=UDP>;tag=6b32c832
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.' Method: ACK

abw1oim · April 12, 2014, 1:47pm

Here You got a different - and cortrect - behaviuor:

[code]Transmitting (NAT) to 63.209.144.201:5060:
ACK sip:navito.martin@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK264d7a97;rport
Max-Forwards: 70
From: “Divito” sip:navito.martin@63.209.144.201;tag=as4879fc5b
To: sip:navito.martin@sip.skype.com;tag=ca90d13f-13c4-53470b44-dde649a1-61fbfd03
Contact: sip:navito.martin@192.168.0.103:5060
Call-ID: 2dd6f23537eebf0409997eb267677633@63.209.144.201
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.8.1
Content-Length: 0

Scheduling destruction of SIP dialog ‘2dd6f23537eebf0409997eb267677633@63.209.144.201’ in 20672 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/ZOI6001-00000002’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (NAT) to 192.168.0.100:58657 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 65.183.8.50:26417;branch=z9hG4bK-d8754z-11d79e29a0357f32-1—d8754z-;received=192.168.0.100;rport=58657
From: "Divito"sip:ZOI6001@192.168.0.103;transport=UDP;tag=6b32c832
To: sip:4321@192.168.0.103;transport=UDP;tag=as4781bb20
Call-ID: NjNjY2VhZGQ0ZDYzNjU5NjVhOGRiMTBlOTk3ZWQ0Nzk.
CSeq: 2 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0[/code]

telss You, that skype accepted Your call but Your Account was busy. This is correct, as You’re trying to reach Yourself.
You need to think about, to whom You want to be connected to via skype and need to adopt Your Skype-Outbound-Context accordingly.

rubenmera · June 5, 2014, 12:28am

try something like this in your dialplan.
/etc/asterisk/extensions.conf

_1NXXNXXXXXX is expecting a number like 18005551212 and will include that it in the variable {EXTEN} to be sent to Skype.

The word Skype is actually the name of my peer in sip.conf [Skype]