Displaying 20 results from an estimated 1000 matches similar to: "REFER mesage extraction using SIP_HEADER"
2010 Nov 23
2
Function SIP_Header not registered
Hello,
I'm trying to use SIP_HEADER function on my dialplan but I receive this
message (on the console):
pbx.c:3367 ast_func_read: Function SIP_Header not registered
Why?
Thank's
- Bakko
2007 Apr 09
3
sip_header=value?
Hi all,
is there anyway i can set SIP_HEADER(To) to the value i like?
--
Regards
Rizwan Hisham
Software Engineer
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2006 Oct 23
0
SIP_HEADER function; what names are available?
Minor update - use the following:
> if (strcasecmp(data,
> "x-Asterisk-Request-URI-pseudo-header")==0)
> {
> ast_copy_string(buf, p->initreq.rlPart2, len);
> -----Original Message-----
> From: Steve Langstaff
> Sent: 23 October 2006 09:58
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users]
2007 Dec 06
2
Print CALLERID in CLI during "pri debug "
Hi all,
I was wondering if it is possible to print the callerid value in the
CLI when doing 'pri debug span 1'
For example
> Call Ref: len= 2 (reference 2707/0xA93) (Terminator)
> Message type: CONNECT (7)
> [18 03 a9 83 97]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0
> ChanSel: Reserved
>
2007 Nov 06
1
Extracting custom headers from SIP REFER
Asterisk 1.4.12
I wish to extract some custom headers from a SIP REFER message but am unable
to do so. However I can extract them from an INVITE. The code is:
exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ;
exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ;
Examples of the INVITE (works) and REFER (doesn't) messages are below.
U 147.202.001.001:5060 ->
2007 Oct 04
5
Setting caller id value on outgoing calls using .call files
Hi all,
I was looking at a way to add the caller id to the outgoing calls (which are
made using .call files) using asterisk. Any ideas how to do this ?
Currently I get 'Unknown' number displayed on my phone when asterisk makes
an outgoing call.
Also using something like this is not working as it still displays unknown
number. I want set the callerid on the 1.call which is made.
exten
2007 Oct 04
1
Asterisk Caller ID Info
Hi Asterisk Users,
I was wondering why a call that is received from Asterisk shows a caller ID
'Unknown' . So here is the scenario,
'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'.
'Asterisk' calls 'B'. 'B' gets joined to the same conference also.
'B' somehow receives the caller ID 'Unknown' and not the
2006 Jun 28
0
Getting at SIP error with SIP_HEADER() ?
Hi,
when attempting to dial an invalid number with Nikotel this is returned:
SIP/2.0 400 Bad Request request uri is neighter my realm or a valid dns
and Asterisk prints smth similar on the CLI. However it appears that I
cannot get access to "400 Bad Request" from the dialplan because this
error is not part of any SIP header, and therefore the function
SIP_HEADER won't do the
2007 Sep 18
2
ISDN PRI debug in Asterisk
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command
" pri intense debug span 1 " , does it debug every packet received
(control and voice/data packets) ?
Thanks
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
2007 Nov 26
2
Get IP address of an incoming or outgoing SIP call
Hi * Users,
What is the way from the dial-plan to get the IP address of an
incoming or outgoing SIP call? I can see the IP address of the SIP
call using 'sip show peers' from the CLI.
Thanks
Regards
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel: 1-646-387-5998
2007 Feb 04
1
Help - Received response: "Forbidden" from '"Unknown"
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Ext No Problems
Panasonic Ext -> SIP Ext No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",
2009 May 17
1
Capture "Server" header in SIP reply.
Hi,
I am trying to capture "Server" header in a 200 OK reply message.
My idea was to use Dail(SIP/user at domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
For example:
[default]
exten => _X.,1,Dial(SIP/user at domain,30,M(GetOtherPartyInfo))
exten => _X.,n,Hangup()
[macro-GetOtherPartyInfo]
exten => s,1,NoOp(SIP Server:
2007 Nov 02
3
use dial plan passed arg value in C agi code
Hello * users,
I know that passing variable in the AGI script is by
exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being
passed and simple_c_prgm is C code
Now how will I receive these variables within C code ? Is it by the same way
arguments are passed in command line to C by using argc and argv or there is
more to be done than that?
Thanks
Regards
--
Arpit Mehta
2017 Jun 05
2
Extensions of sip trunk
Hi,
I just started with setting up a new asterisk system, that will operate on a
sip trunk, but I wonder, how to transfer the calls to different extensions,
because all calls appear as being send to the base number of the trunk.
E.g. given the trunk range of 1234567800-12345678099, a call to 1234567800 is
matched by the same pattern as a call to 12345678099.
; matches 12345678099, too
exten
2007 May 17
2
Call to an arbitrary outbound number by asterisk
Hi,
I have a strange problem. I have a TE110p digium card.
I want to dial 19173995791 when any incoming call comes in. What is
happening is that when I dial 19173-995791. Asterisk picks up the first 5
digits assuming it is the extension and appends 212-85 (here in the
university most numbers start with this) in front . Therefore I get
connected to some random number 212-85-(19173) (where the
2019 Mar 29
3
why doesn't extension "s" work ?
I'm using "s" extension in my dialplan:
[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To)
same=>n,....
But when a call comes in to the gv-voice context, "s" doesn't match the
extension:
res_pjsip_session.c:2991 new_invite: Call from
2016 Mar 18
2
Incoming INVITE with Portability Info and LRN
On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net>
wrote:
> Le 18/03/2016 16:20, Trey Hilyard a ?crit :
> > I am trying to set up my Asterisk server so that it will recognize an
> > incoming call to the Asterisk's own Location Routing Number (LRN),
> > validating the "rn" in the INVITE and then using the Called Number from
>
2007 Oct 07
0
Getting DTMF digits
I forgot to add that this is a T1 ISDN PRI line on which I am sending
the DTMF digits.
Regards
Arpit
On 10/5/07, Arpit Mehta <arpitm at gmail.com> wrote:
> Hi,
>
> Is there any way to get the DTMF digit preferably in the
> extensions.conf . The dtmf digits would be entered by the user
> like"1234567890P1234#" . It doesnt matter whether to put 'P' or
2009 Nov 29
3
Parsing custom SIP headers
Hi,
Just to be sure: Is there a dialplan function in Asterisk that
parses custom "name-addr"-style SIP headers for me?
If I wanted to do it right the syntax
name-addr *(SEMI generic-param)
is quite complex to parse in the dialplan using nothing but CUT().
It's so easy to make false assumtions about angle brackets (< >),
whitespace (LWS), quotes (") around the
2015 Apr 07
5
Asterisk Inbound calls, multiple SIP accounts, calledID
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on.