[asterisk-dev] Another module for testing: chan_console

[Adrià Vidal]

Great, compiled and running

Only a few bugs:


A call to Musiconhold is OK. But imposible to hang up the channel.

A call to a sip endpoint fails about codec....

dial  1001
    -- Executing [1001 at from-internal:1] NoOp("Console/default",
"from-internal") in new stack
    -- Executing [1001 at from-internal:2] Set("Console/default",
"CALLERID(all)=100147") in new stack
    -- Executing [1001 at from-internal:3] Dial("Console/default",
"SIP/adamvozip/1001|30|T") in new stack
[Jan  3 00:18:09] WARNING[91464]: channel.c:610 ast_best_codec: Don't know
any of 0x8000 formats
[Jan  3 00:18:09] WARNING[91464]: channel.c:610 ast_best_codec: Don't know
any of 0x0 formats
Audio is at 192.168.1.103 port 17678
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x40 (slin) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.36.71.100:5060:
INVITE sip:1001|30|T at sip.adamvozip.es SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK61e2ac31;rport
Max-Forwards: 70
From: "100147" <sip:100147 at sip.adamvozip.es>;tag=as442d100c
To: <sip:1001|30|T at sip.adamvozip.es>
Contact: <sip:100147 at 192.168.1.103>
Call-ID: 4c632292789cbb804fabaeb07015b694 at sip.adamvozip.es
CSeq: 102 INVITE
User-Agent: AppleTV PBX
Date: Wed, 02 Jan 2008 23:18:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 391

v=0
o=root 1727718394 1727718394 IN IP4 192.168.1.103
s=Asterisk PBX SVN-trunk-r96025
c=IN IP4 192.168.1.103
t=0 0
m=audio 17678 RTP/AVP 97 3 8 0 10 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:10 L16/8000


Any suggesion?

*CLI> core show channel Console/default
Console/default is not a known channel



*CLI> soft hangup Console/default
Console/default is not a known channel




On Jan 2, 2008 9:39 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:

> Adrià Vidal wrote:
> > have  $ svn co
> http://svn.digium.com/svn/asterisk/team/russell/chan_console
> > dead?
> >
> > trying to test it into my macbook too...
>
> It's been merged into SVN trunk already, you can just test the trunk
> instead.
>
> --
> Kevin P. Fleming
> Director of Software Technologies
> Digium, Inc. - "The Genuine Asterisk Experience" (TM)
>
> _______________________________________________
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-- 
--
Adrià Vidal
adriavidal at gmail.com
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