Hi,
Message type: SETUP (5)
...
XXX Missing handling for mandatory IE 24 (cs0, Channel Identification) XXX
According to ETS 300 102 (the european ISDN specification), section
3.1.16 a SETUP message must contain an IE 'Channel Identification' which
is mandatory for network to user
Tomasz Chmielewski wrote:
exten = _0.,1,Dial(Zap/0/${EXTEN:1})
set g0 instead of 0:
exten = _0.,1,Dial(Zap/g0/${EXTEN:1})
Deti
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To
td wrote:
-- Executing NoOp(Zap/4-1, ) in new stack
-- Executing Dial(Zap/4-1, SIP/tdhome) in new stack
-- Called tdhome
Same problem here. Any ideas?
Deti
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Hi there,
all that started by investigating what happens if SIP clients are
calling anonymously.
The problem: Every client who is registered as a regular user with
username and secret can fake any callerid in subsequent INVITEs.
Asterisk does not apply an accountcode or callerid from sip.conf.
C F wrote:
Welcome to SIP, this is how SIP works, thats why ppl use IAX.
Welcome to SIP for dummies: You have to distinguish between SIP callerid
and authentication. First a callerid is used to call another party or
to identify yourself to another party. Such a callerid is sent via a
This is a preliminary fix for the exploit identified in my last
postings. By far it would be better to fix the find_user call to look
for both, the From-header and an username in the
Proxy-Authorization-header. We even should set a environment variable
(which can be used for dialplans) to
Hi,
how can I setup asterisk to use the number presentation bits on the isdn
side to suppress the number presentation? We need to transmit the
subscriber number for billing purposes via ISDN whether or not the user
wants to hide his/her number. Is there any way to do this?
Deti
Marcello Lupo wrote:
effectively the user is not unknown to the system. He is authenticated with
SIP username and password of a particular peer and he only select to send
anonymous from the phone and if it remain in this way we cannot bill him
currently I'm stuck with the same problem. There
Jason Williams wrote:
show application SetCallerPres
Well I know about that application but if a phone is configured to call
anonymously the Callerid looks like From: Anonymous
sip:[EMAIL PROTECTED] and I found to way to figure out if
this is call originated from an authenticated user. In
Peter Svensson wrote:
What is c-ourcallstate set to at this time? Can you provide a debug log
(pri intense debug span xxx) of the call?
it's Q931_CALL_STATE_ACTIVE - that's what it should be after a call is
established.
Asterisk only expects INFORMATION elements when expecting overlap digits
Peter Svensson wrote:
Ok, then INFORMATION with keypad IE needs to be handled differently from
IE called number.
This is what it looks like with pri intense debug enabled:
Informational frame:
SAPI: 00 C/R: 1 EA: 0
TEI: 000EA: 1
N(S): 116 0: 0
N(R): 126 P: 0
8 bytes of data
--
Hi there,
I tried to use Voicemail from a PRI interface but it didn't work because
pressing a key on the ISDN phone just caused Q931_IE_KEYPAD_FACILITY
messages which are normally handled by a bri-stuffed libpri.
Unfortunately a wrong if condition stops keypad messages from being
passed to
Hi there,
app_mp3 still does not work with the latest bri-stuff patch and the
zaphfc driver. Here in my place it only works with the patch attached.
For me it seems the bri-stuff worsens the asterisk timing... has anybody
else made experiences with it?
Deti
Index: app_mp3.c
Hi there,
I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become
better. I have got lots of dropouts on the IAX2 link (no matter if
jitter buffers are enabled).
Further the MP3Player application does not playback streams like
http://somestreamserver/somestream. It stops saying:
Maurizio Marini wrote:
[controller1]
msn=0xx
...
when i issue an outside call i get:
-- Executing Dial(SIP/sip1-07f4, CAPI/0721xx:bBYEXTENSION:1) in new stack
-- data = 0721xx:b0721950396:1
-- capi request omsn = 0721xx
Aug 2 17:53:02 NOTICE[1224547248]:
Hi there,
I am using bri-stuff.0.0.2 and maybe I misunderstood something but my
HFC card is in bri_cpe_ptmp mode and gets routed about 80 MSNs. Some of
them are not intended to be used by asterisk but every incoming call is
accepted even if the default extension leads to Congestion:
--
Dave Cotton schrieb:
I'm getting this message when I start Asterisk
chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81
but when I try and recompile I get this
chan_capi.c:60: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
undeclared here (not in a
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