Displaying 10 results from an estimated 10 matches for "ruri".
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2006 Mar 06
2
Confusion about construction of RURIs from contact headers for BYEs generated by *
I'm a bit confused about how * constructs the RURI when it generates a
BYE. For the situation where * send the initial INVITE it constructs the
RURI for the BYE from the contact header of the 200 OK response which is
well and good. However when * receives the initial INVITE it does not
use the contact header contained within to construct the BYE...
2007 Jan 05
1
integrating with Asterisk and OpenSER for Voicemail
...In openser.cfg ........... is not hiiting the Asterisk server
............. ... any one help me ........
....
....
modparam("tm","fr_timer",6)
modparam("tm","fr_inv_timer",24)
modparam("tm","wt_timer",1)
#mrodparam("tm", "ruri_matching", 0)
#modparam("tm", "via1_matching", 0)
modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops","avp_ali...
2015 Dec 08
2
host parameter equivalent in pjsip.conf
Hi,
I'm trying to port our configuration form sip to pjsip channel and have
following issue.
Sip.conf has a host parameter that sets the RURI to a given value. This
functionality is needed in some of our scenarios where we need to send
requests to specific IP address with specific domain in RURI.
I did not found an equivalent to the host parameter in pjsip configuration.
Did I miss something?
All I could come with is to get the...
2013 Sep 16
1
asterisk 1.8 sends "SIP/2.0 481 Call/Transaction Does Not Exist" to INVITE
...sdp
Content-Length: 252
Proxy-Authorization: Digest
username="909003660716",realm="X.YYY.32.10",nonce="5237559000011a22ed0fae66765d46ef9131e311fbb9d2fb",uri="sip:8009499014 at X.YYY.32.10
:5060",response="cb6306569b3047ac35064dcb5aee6db4"
X-Enswitch-RURI: sip:8009499014 at X.YYY.32.10:5060
X-Enswitch-Source: X.YYY.33.178:5060
The only problem I see with this INVITE is the VIAs are not right after the
INVITE line... although in https://www.ietf.org/rfc/rfc3261.txt, it
explicitly states the the order of the headers is not a requirement, it
seems A...
2008 Jan 15
0
sip channel error - extension pattern matching problem
...--- SIP read from 192.168.129.38:7160 --->
INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0
Record-Route: <sip:192.168.129.38:7160;lr=on>
...
for instance when I use
such extension:
exten => _vm_.,1,NoOp(-- Context routing-sip-voicemail for ${EXTEN} --)
Asterisk finds extensions for RURI like:
<--- SIP read from 192.168.129.38:7160 --->
INVITE sip:vm_hellboy at ms.sip.rd.touk.pl SIP/2.0
...
Is this an error?
What did I miss off not?
Thanks in advance
Tomasz
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2016 Jul 27
2
Identify endpoint based on Diversion header
...on the Diversion header?
In my scenario, I have some unregistered endpoints (mobile phones) that make calls through our Asterisk, which controls the external call rights based on the endpoint's context. In a normal call, their number will be in the From header and the destination in the To an RURI, but when they make a call forwarding I will receive as calling id a number that it is not known to our system (virtually any number in the world) and my endpoint's number will come in the Diversion header. Asterisk rejects the calls as unauthorized.
Basically I want to allow my users to forwa...
2006 Oct 23
0
SIP_HEADER function; what names are available?
...etc
> > >
> > > I can get Record-Route, Via, From, To etc but don't know
> how to get
> > > the bit after the INVITE. Interestingly only the first Via
> > is returned
> > > by ${SIP_HEADER(VIA)}.
> > >
> > > I've tried R-URI, RURI, URI, ALL, *, blank.
> > >
> > > Any advice appreciated.
> > >
> > > Cameron
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing...
2007 May 10
1
call transfer to asterik.. asterisk as an end point
Hello All.
I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience.
I want to use asterisk for call park/pickup and have configured openser
to relay calls made to ruri 700-720 to asterisk running on
localhost:5069
Call flow:
phone A calls phone B (both phones are polycom)
Phone B answers
then phone b user presses transfer and dials 700
asterisk plays back 701 as the parking lot location
phone B user presses transfer again.
at this time p...
2003 Apr 16
4
iLBC
i tried asterisk ilbc codec against kphone. when the call got
connected, i started to immediately get these kind of message to the
console:
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)?
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of
2012 May 04
4
Interweaving of two datasets
I have two datasets, the first has this shape (each word is a column)
Name address phone .. ..
The second one has the following shape
Name request
I need a contingency table with for example phone and request.
The people registered in these datasets are present in both datasets, BUT in
the first every record is a person, so every person is counted once and is 1
row, in the second every row is