Asterisk <> Sisky (personal edition) problem

Hi all,

Some time ago, I built a small asterisk system for a friend of mine (its using the asterisk "plugin" for IPCOP) - the version is ...er...hmm..wheres PuTTY....ah here it is :

Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q

My Mate wants to use skype as a secondary connection to do least cost routing (now stop rolling those eyes you purists...yes I know...I KNOW...but that's what he wants!)
he also wants skype calls to terminate on his DECT phone.

So on his windows workstation, I have installed the 30day demo of sisky personal edition.

I have setup sisky as an extension and configured asterisk as
follows:

[skypegateway]
type=friend
secret=his_secret
qualify=yes
port=5060
host=dynamic
dtmfmode=rfc2833
dial=SIP/401
context=skypeextn
canreinvite=no
callerid=skypecall

in extensions.conf I have added:

;skype gateway
;incoming calls are routed by sisky to 401(dect phone)
;outgoing calls prefixed by 88 are routed to sisky
[skypeextn]
exten=> 401,1,Dial(ZAP/1)
exten=> 401,2,Busy
exten => _88.,1,Dial(SIP/${EXTEN:0}@skypegateway)

incoming calls (i.e. skype-> asterisk) work perfectly

outgoing calls fail, resulting in an audio glitch followed by a constant tone (eminating I suspect from Sisky) which very occasionally crashes as a result.

I can rule out (no pun intended!) the firewall host's IPTABLES config. There is no NAT between the protected subnet and the firewall host

It's almost certainly a no-brainer to fix..., but I cant see it for looking ! :-/

Any clues ?

Regards

BB

Hi

Couple of things, if you remove the callerid= line fromthe peer you will get the skype name as callerid.

as to the problem you are sending the 88 out as well is that intentional

exten => _88.,1,Dial(SIP/${EXTEN:0}@skypegateway)

what does the sysky log say ?

Ian

Hi Again Ian,

yes the ":0" was a significant part of my blunder <blush> !

However, sisky Personal still wont play.

I made the variable ":2" to lop of the 88,

I set up a shortdial 123 to point to "echo123"

This time I see 123 being sent to sisky (123@skypegateway)in the console.

I see the incoming call at sisky:

2008-02-02 10:05:00 Attempting to connect skype...
2008-02-02 10:05:00 SIP register successfully.
2008-02-02 10:05:00 Skype ID:bigbloke
2008-02-02 10:05:03 24 more days available for trail!
2008-02-02 10:05:03 Skype connect successfully.
2008-02-02 10:07:13 New call from SIP.
2008-02-02 10:07:13 Caller name:"test(PC)"
2008-02-02 10:07:37 Call finished.

I hear what sounds like a dialtone from sisky, so I tried sending 123 to line after hearing it.

as soon as I send dtmf to line sisky crashes citing ntdll.dll

the sip call stays up - looking in the console it has no reason to doubt the call is still in progress.

Anyone seen this behaviour before ?

sisky is running on an athlon dual core 4.2GHz with 2GB of ram so it isnt running out of "grunt" :-/

Regards

BB

Ok for dialing out the format is

exten => _88.,1,Dial(SIP/<siskyexten>${EXTEN:2}@<siskyexten>)

Ive not seen sisky crash like that and I have it here on a lap top

Ian

working now in a fashion Ian, thanks once again

I note that from here, the skype call test service is very poor on downlink after the playback, it keeps dropping in fact

now to get rid of that awful kitaro's brother-in-law MoH! I uploaded a better wav, but sisky is ignoring it for some reason

perhaps i need to register to enable that ?

regards & thanks again

BB

Hi Sorry to say but registering doesnt get rid of it

I think it is embeded in the app, But I can ask the question..

Ian

interpreting the "Chinglish" in the sisky options "system" tab
I read:

Waiting Music (Music of waiting callee answer)

and presumed that this was where you specified the WAV to play whilst waiting for the Skype destination to answer.

I have uploaded a new "skypeout.wav" file and referenced it,
but it seems to ignore that setting, unless that setting is actually for the

autoanswer skype incoming call

option.

although in all honesty, I can't see the purpose of Sisky answering an incoming call, as surely its more important that the SIP endpoint answers it ?

I need to go and look up how many concurrent skype calls sisky PE can handle as I might consider using it myself to redirect incoming skype calls to asterisk, then use asterisk to dial out to my mobile skype account back through sisky for a "single point of calling" option.

Regards

BB

Hi Ill save you the time.

It handles 1 call

the EE edition handles as many as you buy licences for

Ian

ah! ok

need to spend some money then

cheers

BB

$55 per channel

hm....

I can just stomach that given the current exchange rate !

ianplain :

Couple of things, if you remove the callerid= line fromthe peer you will get the skype name as callerid.

having now resolved my ZAP channel incoming CLI issue, I noticed that I dont get the skypename but the sipname as callerid ?

It seems that somehow the old callerid (now removed as you suggested) has been captured by an AGI script which appears to be reusing it (dialparties.agi)

got to be at a "secret facility" south of the m25 tomorrow am so i've had enough huffing about for tonight

Regards

BB

Hi again Ian,

do you get cut off after 42mins and 8 seconds when using sisky <> asterisk ?

Is this a feature of the 30 day demo ?

Regards

BB

(who just had this happen 3 times tonight)