SFA - Ougoing Skype call hangup after 1 ring

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SFA - Ougoing Skype call hangup after 1 ring

Postby herry2003 » Fri Apr 02, 2010 6:20 am

Hi All,

I got problem with outbound call to Skype user via SFA, it just ring 1 time on the opposite Skype client and Hangup.
Asterisk server log show "Everyone is busy/congested at this time (1:0/0/1)"

The Skype client side status is online and "Allow everyone to call me"

There is no issue with incoming Skype call, it can ring the asterisk extension for a while.

Anyone have any cure on this?

Best Regards,
Herry

Here is a copy of my config files and SFA version

*CLI> skype show version
Skype For Asterisk Components:
Channel Driver: 1.6.0_1.0.9.2
Library: 1.6.0_1.0.9.2

***chan_skype.conf***

[sfauser]
secret=XXXXX
context=staff-international
exten=2237
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
direction=both

***sip.conf***

[2236]
type=friend
secret=XXXX
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=2236@device
host=dynamic
dtmfmode=rfc2833
dial=Skype/sfauser@skypeuser
context=staff-international
canreinvite=yes
callerid=device <2236>


**** The output from the asterisk panel ****

-- Executing [s@macro-dial:3] AGI("SIP/2388-00000006", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is '2388' number is '2388'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 2236 to extension map
-- dialparties.agi: Extension 2236 do not disturb is disabled
> dialparties.agi: extnum 2236 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 4
-- dialparties.agi: dbset CALLTRACE/2236 to 2388
-- dialparties.agi: Filtered ARG3: 2236
-- <SIP/2388-00000006>AGI Script dialparties.agi completed, returning 0
Dial("SIP/2388-00000006", "Skype/sfauser@skypeuser,"",tr") in new stack
core.cpp: calling create_control_socket for oid 1139
-- Called sfauser@skypeuser
-- Skype/sfauser-0847b048 is ringing
ACallMember.cpp: Destroying call member 0x84985c0
ending call
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial:8] Set("SIP/2388-00000006", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/2388-00000006", "0?CHANUNAVAIL,1") in new stack
herry2003
Newsterisk
 
Posts: 2
Joined: Fri Apr 02, 2010 4:32 am

Re: SFA - Ougoing Skype call hangup after 1 ring

Postby twilson » Tue Apr 06, 2010 9:36 am

You might check that you have codec_g729 installed, and switch the allow=gsm to allow=ulaw (I don't think SFA supports gsm). Other than that, I'd say contact Digium support so you can safely post some of the debug output that would be required to track this down.
twilson
Oldsterisk
 
Posts: 87
Joined: Mon Jun 01, 2009 9:35 am

Re: SFA - Ougoing Skype call hangup after 1 ring

Postby herry2003 » Thu Apr 08, 2010 11:07 pm

Hi Twilson,

Thanks your advise, I did installed codec_g729, and only allow g729 and ulaw for login in chan_skype

*CLI> g729 show version
Digium G.729A Module Version 1.6.0_3.1.4 (optimized for i686_32)

I conduct further test and suspect the issue on chan_skype side, as sometime it ring 1, sometime 2, which is depends on what Skype client version I call on the opposite site. And if other side pick up the call on the first ring, the call always can establish without issue, which seems no issue on the codec side

Do you know how to get more debug log for SFA, I did enable chan_skype debug, but not much detail provide except the message "== Everyone is busy/congested at this time (1:0/0/1)" which terminate the call

*CLI> skype set debug on
herry2003
Newsterisk
 
Posts: 2
Joined: Fri Apr 02, 2010 4:32 am


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