[asterisk-dev] Re: asterisk-dev Digest, Vol 22, Issue 25

Leke Fasola lekefasola at gmail.com
Wed May 10 00:28:13 MST 2006


Thanks

On 5/9/06, asterisk-dev-request at lists.digium.com <
asterisk-dev-request at lists.digium.com> wrote:
>
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>
> Today's Topics:
>
>   1. Re: outbound proxy (Kevin P. Fleming)
>   2. Re: outbound proxy (Mikael Magnusson)
>   3.   Re: ISUP over MGCP and/or SIP - A Project and, BOUNTY
>      (Ahmed Naji)
>   4. Re:  Re: ISUP over MGCP and/or SIP - A Project and,       BOUNTY
> (Anton)
>   5. There is Set, but is there Get? (Peter Beckman)
>   6. Hardwareless PCI Card Simulator (Hugo Saporetti Junior)
>   7. Re: There is Set, but is there Get? (Tilghman Lesher)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 09 May 2006 04:12:03 -0500
> From: "Kevin P. Fleming" <kpfleming at digium.com>
> Subject: Re: [asterisk-dev] outbound proxy
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <44605CE3.1040408 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Raymond Chen wrote:
> > I mean is the 2859 patch in the bug list
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002859, did it ever
> merge to
> > the current 1.2 version or trunk?
>
> It's not noted in the bug notes, so I don't know that we know the answer
> to that question.
>
> Again, describe (on the proper mailing list, not this one) what you are
> trying to accomplish and somebody can help you determine whether it is
> supported or not.
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 09 May 2006 12:39:13 +0200
> From: Mikael Magnusson <mikma264 at gmail.com>
> Subject: Re: [asterisk-dev] outbound proxy
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <44607151.3000401 at gmail.com>
> Content-Type: text/plain; charset=us-ascii; format=flowed
>
> Raymond Chen wrote:
> > I mean is the 2859 patch in the bug list
> > http://bugs.digium.com/bug_view_page.php?bug_id=0002859, did it ever
> merge to
> > the current 1.2 version or trunk?
> >
> > thanks
> >
> > Ray
> >
>
> "Initial per-peer in there (no docs yet) but not global outbound proxy."
> (Note: 0017897)
>
> Commited in r4383.
>
> Mikael
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 09 May 2006 13:21:54 +0100
> From: Ahmed Naji <worldentropy at yahoo.co.uk>
> Subject: [asterisk-dev]         Re: ISUP over MGCP and/or SIP - A Project
>        and, BOUNTY
> To: asterisk-dev at lists.digium.com
> Message-ID: <44608962.5090807 at yahoo.co.uk>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> 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 win FIFA World Cup
> > tickets. http://uk.mail.yahoo.com
> >
> >
> > ------------------------------
> >
> > 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.
>
>
> ___________________________________________________________
> Switch an email account to Yahoo! Mail, you could win FIFA World Cup
> tickets. http://uk.mail.yahoo.com
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 9 May 2006 17:28:29 +0500
> From: Anton <anton.vazir at gmail.com>
> Subject: Re: [asterisk-dev]  Re: ISUP over MGCP and/or SIP - A Project
>        and,    BOUNTY
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <200605091728.29463.anton.vazir at gmail.com>
> Content-Type: text/plain;  charset="iso-8859-1"
>
> 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
> > > win FIFA World Cup tickets. http://uk.mail.yahoo.com
> > >
> > >
> > > ------------------------------
> > >
> > > 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 $5,000 for a documented,
> > >> demostrable version, with further bounties put on
> > >> mile-stones.
> > >
> > > I think bounty is too low.
> > >
> > > Also, possibly, a better idea is to enhance chan_ss7 to
> > > support (instead of zaptel) the SIGTRAN stack (from
> > > openss7.org). SCTP *is* a standardized protocol after
> > > all...
> >
> > Hi Alex,
> >
> > Well .. SIGTRAN enhancement to chan_ss7 is indeed a good
> > idea. Reference the bounty, I don't particularly think
> > it's too low, as this is only for the first mile-stone.
> >
> > Thanks,
> >
> > Ahmed.
> >
> >
> > _________________________________________________________
> >__ Switch an email account to Yahoo! Mail, you could win
> > FIFA World Cup tickets. http://uk.mail.yahoo.com
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 9 May 2006 13:17:52 -0400 (EDT)
> From: Peter Beckman <beckman at purplecow.com>
> Subject: [asterisk-dev] There is Set, but is there Get?
> To: asterisk-dev at lists.digium.com
> Message-ID: <20060509131136.R7824 at thermonuclear.org>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> In documenting things, I'm realizing that many "Set..." commands were
> replaced by "Set(varname=value)."  However, there doesn't seem to be a
> "Get" command to match.  I assume this is because you can reference
> everything as a variable that you set.
>
> So, effectively, AppendCDRUserField() is equivalent to:
>
>     Set(CDR(userfield)=${CDR(userfield)}value)
>
> Please correct me if I'm wrong, or confirm that I'm right.
>
> Is the prefered usage to use the AppendCDRUserField function, or to use
> the
> Set() application?  Will AppendCDRUserField be deprecated at some point?
>
> Beckman
>
> ---------------------------------------------------------------------------
> Peter Beckman                                                  Internet
> Guy
> beckman at purplecow.com
> http://www.purplecow.com/
>
> ---------------------------------------------------------------------------
>
>
> ------------------------------
>
> Message: 6
> Date: Tue, 9 May 2006 14:28:51 -0300
> From: Hugo Saporetti Junior <hugos at inatel.br>
> Subject: [asterisk-dev] Hardwareless PCI Card Simulator
> To: asterisk-dev at lists.digium.com
> Message-ID:
>        <18E6815BB347BE43894F742F38D0F8CA0441DDE2 at hertz.local.inatel.br>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello
>
>
>
>            I would like to know if there's  any software to simulate a PCI
> WCFXO card. If so, where could I download it? I want to use it with
> Zaptel/Asterisk. Regards
>
>
>
> _____
>
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> ------------------------------
>
> Message: 7
> Date: Tue, 9 May 2006 13:15:20 -0500
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [asterisk-dev] There is Set, but is there Get?
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <200605091315.21510.tilghman at mail.jeffandtilghman.com>
> Content-Type: text/plain;  charset="iso-8859-1"
>
> On Tuesday 09 May 2006 12:17, Peter Beckman wrote:
> > In documenting things, I'm realizing that many "Set..." commands were
> > replaced by "Set(varname=value)."  However, there doesn't seem to be
> > a "Get" command to match.  I assume this is because you can reference
> > everything as a variable that you set.
>
> That is correct.
>
> > Is the prefered usage to use the AppendCDRUserField function, or to
> > use the Set() application?  Will AppendCDRUserField be deprecated at
> > some point?
>
> Set, and yes.
>
> --
> Tilghman
>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> End of asterisk-dev Digest, Vol 22, Issue 25
> ********************************************
>
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