[asterisk-ss7] RE: ISUP via Media Gateways: A project and possible
bounty
Ahmed Naji
worldentropy at yahoo.co.uk
Sat May 13 14:23:35 MST 2006
Luciano,
You are missing the point. It is exactly because the Cisco AS53xx can't
process ISUP we are doing this. I agree, SIP-T is not adequate because
it misses some information, although it is possible to encapsulate the
info and send it via other means (Cisco GKApi for example), but this is
not the aim.
The way it is to be done is to configure the gateway's SCTP features to
do this; this will backhaul whatever signalling is present on the
prescribed channel(s) unchanged to a CA, including custom signalling -
e.g. QSIG ..etc. There is plenty documentation on this, and I personally
have done this with the AS53xx for a variety of signalling setups.
Now we come to the PGW: the biggest, greatest, most broken con in the
soft-switch arena; have I said enough ? :)
Thanks,
Ahmed.
asterisk-ss7-request at lists.digium.com wrote:
> Send asterisk-ss7 mailing list submissions to
> asterisk-ss7 at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-ss7
> or, via email, send a message with subject or body 'help' to
> asterisk-ss7-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-ss7-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-ss7 digest..."
>
>
> Today's Topics:
>
> 1. RE: ISUP via Media Gateways: A project and possible bounty
> (Leonti Dailis)
> 2. RE: RE: ISUP via Media Gateways: A project andpossible bounty
> (Luciano Ramos)
> 3. Re: Some sessions hangs. Is it possible with SS7 (Storer, Darren)
> 4. Re: Some sessions hangs. Is it possible with SS7 (Anton)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 12 May 2006 12:38:13 -0700
> From: Leonti Dailis <Leonti.Dailis at SS8.com>
> Subject: [asterisk-ss7] RE: ISUP via Media Gateways: A project and
> possible bounty
> To: "'asterisk-ss7 at lists.digium.com'" <asterisk-ss7 at lists.digium.com>
> Message-ID:
> <CCFACC5ADB1B1C4CA74C864D47364BC90729668E at ss8mail2.ss8.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
>
>> is that whether or not the SIP code within Asterisk can
>> handle 3rd part
>> call control; if this is the case, then life is much easier:
>> the Cisco
>> can be controlled as a UA, encapsulating the ISUP messages in SIP
>> envelops sent over to Asterisk.
>>
>
> I don't believe you can do it (use SIP-T encapsulation?) without using Cisco
> PGW. The gateway can not process ISUP on its own. You will have to use
> SIGTRAN.
>
> Best regards,
>
> Leonti Dailis
> Principal Engineer
> SS8 Networks, Inc.
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060512/eb4fef41/attachment-0001.htm
>
> ------------------------------
>
> Message: 2
> Date: Fri, 12 May 2006 17:54:39 -0300
> From: "Luciano Ramos" <lramos at telviso.com.ar>
> Subject: RE: [asterisk-ss7] RE: ISUP via Media Gateways: A project
> andpossible bounty
> To: <asterisk-ss7 at lists.digium.com>
> Message-ID: <E1FeefP-00065p-Ru at mail.telviso.com.ar>
> Content-Type: text/plain; charset="us-ascii"
>
> Also, you can't use SIP-T to interconnect the AS53xx to the Asterisk box,
> basically because SIP-T doesn't carry all the information from the lower
> layers of the SS7 Stacks (eg, CIC number, etc.. )
>
>
>
> Luciano
>
>
>
> _____
>
> From: asterisk-ss7-bounces at lists.digium.com
> [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Leonti Dailis
> Sent: Viernes, 12 de Mayo de 2006 04:38 p.m.
> To: 'asterisk-ss7 at lists.digium.com'
> Subject: [asterisk-ss7] RE: ISUP via Media Gateways: A project andpossible
> bounty
>
>
>
>
>
>
>> is that whether or not the SIP code within Asterisk can
>> handle 3rd part
>> call control; if this is the case, then life is much easier:
>> the Cisco
>> can be controlled as a UA, encapsulating the ISUP messages in SIP
>> envelops sent over to Asterisk.
>>
>
> I don't believe you can do it (use SIP-T encapsulation?) without using Cisco
> PGW. The gateway can not process ISUP on its own. You will have to use
> SIGTRAN.
>
> Best regards,
>
> Leonti Dailis
> Principal Engineer
> SS8 Networks, Inc.
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060512/3a83ab07/attachment-0001.htm
>
> ------------------------------
>
> Message: 3
> Date: Sat, 13 May 2006 00:01:47 +0100
> From: "Storer, Darren" <darren.storer at gmail.com>
> Subject: Re: [asterisk-ss7] Some sessions hangs. Is it possible with
> SS7
> To: asterisk-ss7 at lists.digium.com
> Message-ID:
> <1142a46a0605121601w7df5c467xbc7574d95a36fb79 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Anton,
>
> the state of CICs can become unsynchronised between SS7 nodes although it
> doesn't happen too often. Sometimes Node A thinks that a particular CIC is
> "In Service Free" whilst Node B thinks that Node A didn't handle the last
> call correctly and places the same CIC into a blocked state at a higher
> application layer.
>
> When this situation occurs we normally use MML to "flex" the CICs by
> manually taking each suspect CIC out of service and then returning it back
> to service. I have seen that chan_ss7 appears to support these maintenance
> messages but I haven't used them myself.
>
> When an SS7 node is started for the first time (cold start) it has no
> knowledge of the state of the CICs as maintained by another SS7 node that it
> is connected to. To synchronise the state of the CICs after a cold start it
> is common to see many block and unblock messages (including Circuit Group
> Resets) after the signalling links are aligned but before the route starts
> to handle live traffic.
>
> Regards
>
> Darren
>
>
> On 12/05/06, Anton <anton.vazir at gmail.com> wrote:
>
>> Guys,
>>
>> Does anyone know, if it is possible for timeslots to hang, with SS7
>> signalling?
>> For example if REL message is missing (did not received in time by
>> Asterisk) -
>> would the accured timeslot hang (be active on Asterisk side, but
>> released in the TELCO side?) I've strange issues with one of my telco's.
>> The sessions just timed out, and released by the Asterisk itself, since
>> there
>> is 3600s duration limit.
>>
>> Any help is highly appreciated!
>>
>> ,"SS7/15","IAX2/axsoftsw1-16389","Hangup","","2006-04-21
>> 23:04:54","2006-04-21 23:05:00","2006-04-22
>> 00:04:59",3605,3599,"ANSWERED","DOCUMENTATION"
>> ,"SS7/19","IAX2/axsoftsw1-16392","Hangup","","2006-04-21
>> 23:05:23","2006-04-21 23:05:29","2006-04-22
>> 00:05:28",3605,3599,"ANSWERED","DOCUMENTATION"
>> ,"SS7/11","IAX2/axsoftsw1-16393","Hangup","","2006-04-21
>> 23:46:00","2006-04-21 23:46:07","2006-04-22
>> 00:46:07",3607,3600,"ANSWERED","DOCUMENTATION"
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-ss7 mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>>
>
>
>
>
Send instant messages to your online friends http://uk.messenger.yahoo.com
More information about the asterisk-ss7
mailing list