[asterisk-dev] Another module for testing: chan_console

[Adrià Vidal]

On Jan 4, 2008 2:19 PM, Russell Bryant <russell at digium.com> wrote:

> Adrià Vidal wrote:
> > And about the codecs problem?
>
> I'm not exactly sure what the failure was.  I see the couple of warning
> messages, but then I also see that it looks like a valid INVITE got sent
> out.
> What is the rest of the SIP dialog?



OK without or T option on dial

*CLI> console 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") in new stack
[Jan  5 00:47:40] WARNING[22257]: channel.c:610 ast_best_codec: Don't know
any of 0x8000 formats
[Jan  5 00:47:40] WARNING[22257]: channel.c:610 ast_best_codec: Don't know
any of 0x0 formats
    -- Called adamvozip/1001
[Jan  5 00:47:40] WARNING[22257]: channel.c:3329
ast_channel_make_compatible_helper: No path to translate from
SIP/adamvozip-00841000(0) to Console/default(32768)
[Jan  5 00:47:40] WARNING[22257]: channel.c:2957 set_format: Unable to find
a codec translation path from ilbc to unknown
[Jan  5 00:47:40] WARNING[22257]: channel.c:2957 set_format: Unable to find
a codec translation path from ilbc to unknown
    -- SIP/adamvozip-00841000 answered Console/default
  --- <("<) --- Call from Console has been Answered --- (>")> ---


wrong with tT option on dial


The problem seemt the Tt and timeout options send into the SIP header to:

From: "100147" <sip:100147 at sip.adamvozip.es>;tag=as5832ffa0
To: <sip:1001|300|tT at sip.adamvozip.es>;tag=
55945f39a9b6124c308bef45a8f6126d.2cc4
Call-ID: 0a2ec79a6d28c3746409425a1453394b at sip.adamvozip.es



--
Adrià Vidal
adriavidal at gmail.com
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