RFC 3435 is the MGCP protocol specification !!!
Dial Plan is a application layer function , usually at the Call Processing
layer.
SIP as a signaling protocol does not impose nor should dial plans.
Cheers
PS: http://en.wikipedia.org/wiki/Telephone_numbering_plan
On Thu, Oct 14, 2010 at 7:29 AM
Hello
Try pjsip
pjsip.org
Cheers
Sent from my iPhone
On Jul 13, 2010, at 2:27 AM, Vinod Parameswaran
wrote:
> Hello list,
>
> As part of my telephony project on the Unix platform, I need to make use of
> SIP signaling.
>
> Since we are trying something new at the protocol level itself
IMHO, there is now way to guarantee 1:1 mapping due to internetworking ,
have you looked at Q.1912.5 spec ??, ITU defined and more detailed than
3398.
the problem to me is from ISUP to SIP always there is a clearer mapping,
but from SIP to ISUP is not the clear so you have to make educated decisi
you can look into Radvision as well.
On Wed, Feb 3, 2010 at 11:40 AM, Badri wrote:
> Aricent SIP stack.
>
>
>
>
> From: Premalatha Kuppan
> To: sip-implementors@lists.cs.columbia.edu
> Sent: Wed, February 3, 2010 5:21:12 PM
> Subject: [Sip-implementors] Commer
Also
to consider is if the provisional responses have different To tags,
i.e. they are coming from different endpoints (proxy could have
forked) , in which case you should honor whatever provisional response
your are getting until
you get the final 200OK
Cheers
On Oct 25, 2009, at 11:59 PM
Please refer to http://www.ietf.org/rfc/rfc3262.txt
thanks
On Tue, Jun 30, 2009 at 11:48 AM, soma bhargava <
soma.bharg...@globaledgesoft.com> wrote:
> Hi All,
>
> What is the behaviour of UAC in following scenario:
>
> 1. UAC sends INVITE with 100rel in both requires and supported headers.
> UA
check this web site.
http://www.tech-invite.com/
hope it helps
Cheers
On Tue, Jun 2, 2009 at 2:59 AM, Avasarala Ranjit-A20990 wrote:
> Hi
>
> You can refer to these links
>
> SIP-H323 interworking: RFC 4123
> SIP-QSIG interworking: RFC 4497
>
> Regards
> Ranjit
>
> -Original Message-
The original dialed number is supposed to be carried in a diversion
header.
for example say :
A calls B
B forwards to C
C sends you an invite
you should received an invite with a diversion like
Diversion: reason=unconditional ,; some other parameters
Diverssion: ;
also you can get one Divers
My 2 cents
First sending multiple 180 with the same tags is completely valid.
Second, IMO a 183 would be a good answer for this. the only thing to be
careful here is if whether the voice has been cut already or not by MGW.
if it has not, then 183 wo/SDP should be good enough
if the voice has bee
As mentioned here SIP does not provided in itself billing related
mechanism,
however all the information necessary is at hand. Usually the B2BUA would
gather
the critical information to formulate a Call Detailed Record that is to be
sent to an external
machine. This is normally done via some sta
Also
even though this is expired by noe draft. it was a good guideline. i know
some vendors had used it.
http://www.softarmor.com/wgdb/docs/draft-kotar-sipping-dss1-sip-iw-01.txt
thanks
alejandro
On Fri, Apr 17, 2009 at 3:38 AM, Meir Leshem
wrote:
> Hi Padmaja,
> I would suggest to look on EC
Also look at
http://www.faqs.org/rfcs/rfc3515.html
Cheers
On Mon, Mar 23, 2009 at 4:34 AM, Somesh S. Shanbhag wrote:
> You can have look at http://tech-invite.com/Ti-sip-service-4.html
>
> Somesh
>
> * Please do not take print out of this e-mail unless its absolutely
> necessary *
>
>
>
> --
please read sip-t related rfc.
http://www.packetizer.com/rfc/rfc3372
http://www.faqs.org/rfcs/rfc3204.html
Basically you could encapsulate qsig in sip messages containing MIME body
following the above rfc's guidelines.
Cheers
alejandro
On Mon, Mar 2, 2009 at 6:57 AM, Ankit Agarwal
wrote:
>
Try
Eyebeam , Kapanga.
thanks
On Jan 2, 2009, at 7:17 AM, hagai sela wrote:
> Hi,
>
> I am looking for a windows compatible softphone with support for the
> EVRC
> codec, preferably a free one. Does anybody know about such softphones?
>
>
>
> Thanks,
>
> Hagai.
>
>
RADIUS also can be used.
Rgds
Alejandro
On Dec 19, 2008, at 12:10 AM, wrote:
> Billing is made based on operator or service providers if you like to
> read standards you can see some 3gpp documents Ro(online charging) and
> Rf (offline) Rx(policy and charging control) Diameter Interfaces has
> b
see my comments inline
regards
alejandro
On Mon, Dec 1, 2008 at 7:18 AM, Padmaja <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> This is regarding the local ringback tone generation vs playing early media
> as per RFC 3960. I am, not clear on a few points-
>
> 1. When should the Source UA start playing
For the diversion header details. please read:
draft-levy-sip-diversion-08.txt
It is definitely possible that the counter is greater than 1. For example if
the call came in on SS7 the counter is taken from the ISUP redirecting info
parameter. and indicates how many times the call has been forwar
All,
in the following call flow,
is it valid for endpoint A to send a ACK with SDP?
endpointAGW
INVITE w/SDP (offer)-->
<100 trying---
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