Asterisk and Skype Connect Issue

Question

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:

[general]
;register => gnext.telephony:[email protected]/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:[email protected]/99051000234871
register =>99051000234871:[email protected]/99051000234871
;register => 9905100xxxxxxx:[email protected]/9905100xxxxxxxxx
trustrpid=no
sendrpid=yes
calllimit=4
defaultexpiry=240


[skype]
type=friend
;type=peer
;context=from-trunk
context=from-trunk
username=9905xxxxxxx
secret=xxxxxxxx
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

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 '[email protected]:5060'                                          

Method: OPTIONS
[Apr  8 22:26:18] NOTICE[12076]: chan_sip.c:15059 sip_reregister:-- Re-registration        

for  [email protected]
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:[email protected]>;tag=as22c0d1e3
To: <sip:[email protected]>
Call-ID: 53a73ba5083511cb29b1d3cf6bf4f37b@[::1]

CSeq: 109 REGISTER User-Agent: Asterisk PBX 11.8.1 Expires: 120 Contact: Content-Length:

any ideas? Please let me know. Thanks!!

James James · Accepted Answer · 2014-05-15T15:49:36

I hope you've changed your password :)

You should X out your 9905# and the password above (56w2t5EAeCfnum) every time it appears and change your password in Skype Manager. Otherwise someone can use up your outbound call credits or make international calls on your behalf.

I have the same issue with Skype Connect as well working with Asterisk. Inbound calls work fine; outbound calls to my cell don't but I can call some other numbers.

I get the exact same messages in the CLI, that the SIP trunk is circuit-busy.

Try calling to some other numbers and see if you can correlate it to certain providers. I also have a SIP trunk through another provider (I am planning to shut that one down IF I get Skype working reliably).

So far, my impression is Skype is NOT a good bet for SIP trunking. I hope I can get it resolved with them and change that impression. I'm supposed to be hearing from their Level 3 support soon.

Asterisk and Skype Connect Issue

2 Answers