[asterisk-ss7] My experiences with chan_ss7, appreciation,
a question and clarification
ADEGOKE ARUNA
goksie at gmail.com
Mon Mar 20 01:27:55 MST 2006
Thank you, Vazir, Militzer and all other friends,
I got it working on my first ss7 link.
My provider just reset their ss7 and the link comes up and the rbt is ok.
The only thing am seeing now is as stated below.
My status is ok
linkset siuc, link l1, schannel 16, INSERVICE, rx: 1, tx: 3/3,
sentseq/lastack: 35/35, total 7196960, 7197024
I can make call.
However, I need to set up numbers on my 120 usable channels so that when
call hit asterisk from my sip and iax callers the numbers on the e1 ss7
channels will be displaying instead of the unrecongnised number the sip and
iax users are using.
Kindly advise on the logs am seeing and the way out.
Each time I dialed a number 0 is added to my caller id. Why?
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=33.
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=49.
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=65.
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=81.
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=97.
Mar 20 09:02:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=113.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=33.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=49.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=65.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=81.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=97.
Mar 20 09:02:42 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=113.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=33.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=49.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=65.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=81.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=97.
Mar 20 09:03:12 NOTICE[4922]: chan_ss7.c:1066 t22_timeout: T22 timeout (No
"circuit group reset acknowledge" from peer) CIC=113.
Goksie
-----Original Message-----
From: asterisk-ss7-bounces at lists.digium.com
[mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Anton
Sent: Saturday, March 18, 2006 5:49 AM
To: asterisk-ss7 at lists.digium.com
Subject: Re: [asterisk-ss7] My experiences with chan_ss7,some questions and
a solution for the ringback tone
Leonimar, :) I've been posting that in the mailing list and
voip-info.org my patch FIXES that - because I did the same
protocol analysis and I've written about in voip-info -
that is a difference of the ITU and ANSI ISUP
Look at the erricson presentation saying about that.
http://www.google.com/search?q=von9909_ericsson.ppt&sourceid=mozilla&start=0
&start=0&ie=utf-8&oe=utf-8
von9909_ericsson.ppt
On 17 March 2006 20:06, leonimar cape wrote:
> Hi everyone,
>
> Still with regards to the RBT. After attaching a
> protocol analyzer, I found out that asterisk is not
> sending ACM after receiving the IAM from the Nortel
> DMS switch. Also I notice "unknown state 0x03" on the
> channel. What does this mean. Can anyone help. Is this
> message already included in the new release version?
> Nonetheless, I have not problem with calls orginating
> from the asterisk to the pstn side.
>
> Regards,
>
> Leonimar Cape
>
> --- Are <london3 at gmail.com> wrote:
> > Hi
> >
> > I fully support the Timeout parameter as this is a
> > common practice in SIP
> > based communication.
> >
> > I work a lot with Patton SmartNode Sip Gateways and
> > in the configuration we
> > have the following.
> >
> > context cs switch
> > digit-collection timeout 3
> > routing-table called-e164 TEST1
> > route .T dest-interface IF_E1
> > route 00.% dest-interface IF_E2
> >
> > On many SIP phones you also have the option to
> > choose Timeout or *Early
> > Dial *(484 response)
> >
> > You are not fully aware of your call routes in many
> > Real life SIP
> > applications. We all know that International
> > numbering plans are no easy
> > beasts.
> >
> > --
> > Are Casilla
> > http://astartelecom.com - Independent VOIP Telecoms
> > Broker. Asterisk
> > Consultants
> > http://astbill.com - Open Source Billing, Routing
> > and Management software
> > for Asterisk and VOIP
> > AstBill DEMO: http://demo.astbill.com
> >
> > On 3/16/06, Kai Militzer <km at westend.com> wrote:
> > > Hello Jacob, hello all,
> > >
> > > Jacob Tinning wrote:
> > > > We didn't like the timer-solution because we
> >
> > think its wrong to delay
> >
> > > all calls
> > >
> > > > X seconds just because the SS7-asterisk doesn't
> >
> > know another Asterisk's
> >
> > > dialplan.
> > >
> > > Thats why I made it configurable, so that it can
> >
> > be turned off, when not
> >
> > > needed. ;)
> > >
> > > > My suggestions is
> > > > 1. Use identical dialplans on the SS7-gateway
> >
> > and the SIP server
> >
> > > > 2. Store the dialplan in a shared database.
> > > > 3. I think it is (maybe) posible to 'share' the
> >
> > dialplan through IAX
> >
> > > (anybody ?)
> > >
> > > Your suggestions are reasonable if you know the
> >
> > dialplan. In my case it
> >
> > > can be possible that I will forward a number block
> >
> > to a customer. I have
> >
> > > not (and will not have) any knowledge of the
> >
> > length of the numbers the
> >
> > > customer uses, I only know the base of the block,
> >
> > neither does the
> >
> > > customer have to use an asterisk as termination.
> > >
> > > Example:
> > > I have a block +49-241-9909888 [0-99999]. I
> >
> > forward this block to a
> >
> > > customer. This customer can add one to five digits
> >
> > to this block
> >
> > > depending on his needs and I will never have
> >
> > knowledge of how many
> >
> > > digits he uses.
> > >
> > > As you see, if you want use chan_ss7 as a
> >
> > multi-customer SS7-to-SIP
> >
> > > gateway with a national numbering plan without
> >
> > fixed length numbers (as
> >
> > > in the US) there is no way around a timer. It's
> >
> > sad but true. ;)
> >
> > > >>And last but not least, I also had the problem
> >
> > that no ringback tones
> >
> > > >>were generated by asterisk. The following two
> >
> > lines in the dialplan
> >
> > > >>inserted before the Dial statement do the trick:
> > > >>
> > > >>
> > > >>exten => _X.,n,SetLanguage(de)
> > > >>exten => _X.,n,Playtones(ring)
> > > >
> > > > We actually tried this, but we had to insert a
> >
> > ,1,Answer before the
> >
> > > Playtones command.
> > >
> > > > ...but the Answer before Playtones, breaks most
> >
> > telcos billing system,
> >
> > > > since a call is 'from the Answer to a hangup'.
> > >
> > > It works here without the answer as there is
> >
> > early-Media after receiving
> >
> > > an IAM. This works also with MOH instead of the
> >
> > ringback beeps, what can
> >
> > > be quite funny.
> > >
> > > Best regards,
> > > Kai
> > >
> > > --
> > > Kai Militzer WESTEND GmbH |
> >
> > Internet-Business-Provider
> >
> > > Technik CISCO Systems Partner
> >
> > - Authorized Reseller
> >
> > > Lütticher Straße 10
> >
> > Tel 0241/701333-14
> >
> > > km at westend.com D-52064 Aachen
> >
> > Fax 0241/911879
> >
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