[asterisk-dev] Re: ISUP over MGCP and/or SIP - A Project and,
BOUNTY
Anton
anton.vazir at gmail.com
Tue May 9 05:28:29 MST 2006
asterisk-ss7 list is quite active now. Why not to post/ask
it there?
On 9 May 2006 17:21, Ahmed Naji wrote:
> asterisk-dev-request at lists.digium.com wrote:
>
> <SNIP>
>
> > Today's Topics:
> >
> > 1. ISUP over MGCP and/or SIP - A Project and BOUNTY
> > (Ahmed Naji) 2. Re: ISUP over MGCP and/or SIP - A
> > Project and BOUNTY (alex at pilosoft.com)
>
> <SNIP>
>
> > -------------------------------------------------------
> >---------------
> >
> > Message: 1
> > Date: Tue, 09 May 2006 00:37:56 +0100
> > From: Ahmed Naji <worldentropy at yahoo.co.uk>
> > Subject: [asterisk-dev] ISUP over MGCP and/or SIP - A
> > Project and BOUNTY
> > To: asterisk-dev at lists.digium.com
> > Message-ID: <445FD654.2090109 at yahoo.co.uk>
> > Content-Type: text/plain; charset=ISO-8859-1;
> > format=flowed
> >
> > List,
> >
> > I need to implement ISUP functionality by utilising
> > Asterisk as a Call Agent (3PCC in case of SIP), and one
> > ore more Cisco AS5350 or similar as media gateways.
> >
> > The setup will be such that F-Links will be extracted
> > from the PRI circuit(s) and handed over to Asterisk for
> > call processing. The interface would be implemented as
> > a channel (e.g. chan_mgcpisup), much the same way as
> > chan_ss7 is implemented. Where implementation is via
> > SIP, Asterisk would implement 3rd Part Call Control to
> > get the gateway to talk ISUP, and handle the signalling
> > payload in a SIP-enveloped message. The latter (i.e.
> > SIP implementation is preferred).
> >
> > I am thinking to handover the signalling in an H.225
> > payload, process it and hand it over back to the
> > gateway the same way it came: i.e. SCCP/MTP3/MTP2
> > ..etc.
> >
> > Bounty wise, I am thinking $5,000 for a documented,
> > demostrable version, with further bounties put on
> > mile-stones.
> >
> > Thanks,
> >
> > Ahmed Naji
> > 5G Networks LTD
> >
> >
> > _______________________________________________________
> >____ Switch an email account to Yahoo! Mail, you could
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> >
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Mon, 8 May 2006 20:06:34 -0400 (EDT)
> > From: alex at pilosoft.com
> > Subject: Re: [asterisk-dev] ISUP over MGCP and/or SIP -
> > A Project and BOUNTY
> > To: Asterisk Developers Mailing List
> > <asterisk-dev at lists.digium.com> Message-ID:
> > <Pine.LNX.4.44.0605082003290.7535-100000 at bawx.pilosoft.
> >com> Content-Type: TEXT/PLAIN; charset=US-ASCII
> >
> > On Tue, 9 May 2006, Ahmed Naji wrote:
> >> The setup will be such that F-Links will be extracted
> >> from the PRI circuit(s) and handed over to Asterisk
> >> for call processing. The interface would be
> >> implemented as a channel (e.g. chan_mgcpisup), much
> >> the same way as chan_ss7 is implemented. Where
> >> implementation is via SIP, Asterisk would implement
> >> 3rd Part Call Control to get the gateway to talk ISUP,
> >> and handle the signalling payload in a SIP-enveloped
> >> message. The latter (i.e. SIP implementation is
> >> preferred).
> >>
> >> I am thinking to handover the signalling in an H.225
> >> payload, process it and hand it over back to the
> >> gateway the same way it came: i.e. SCCP/MTP3/MTP2
> >> ..etc.
> >>
> >> Bounty wise, I am thinking $5,000 for a documented,
> >> demostrable version, with further bounties put on
> >> mile-stones.
> >
> > I think bounty is too low.
> >
> > Also, possibly, a better idea is to enhance chan_ss7 to
> > support (instead of zaptel) the SIGTRAN stack (from
> > openss7.org). SCTP *is* a standardized protocol after
> > all...
>
> Hi Alex,
>
> Well .. SIGTRAN enhancement to chan_ss7 is indeed a good
> idea. Reference the bounty, I don't particularly think
> it's too low, as this is only for the first mile-stone.
>
> Thanks,
>
> Ahmed.
>
>
> _________________________________________________________
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