If the DMS100 switch can talk signalling directly with Asterisk, without
an STP, it should be possible to use a single timeslot for ss7
signalling, so with 2 T1 you could have 47 voice calls and one
signalling channel. This is common with E1 setups. Also with E1 its
common for a timeslot to be
I am the IP guy.
So basically if I understand you properly, I should be able to do the
SS7+T1 and get proper operation, provided the configuration on both
sides is right.
On Wed, Dec 7, 2011 at 1:06 PM, Marcelo Pacheco wrote:
If the DMS100 switch can talk signalling directly with Asterisk, witho
ly if I understand you properly, I should be able to do the
> > SS7+T1 and get proper operation, provided the configuration on both
> > sides is right.
> >
> > On Wed, Dec 7, 2011 at 1:06 PM, Marcelo
Pacheco wrote:
> >> If the DMS100 switch can talk signalling directl
. SIGTRAN access
didn't offer any cost advantages to operators in A link markets.
-Paul
On Dec 9, 2011, at 3:57 PM, Marcelo Pacheco wrote:
Typical North America SS7 signaling links use a dedicated v.35 link. STPs
and switches come with V.35 interfaces for signaling instead of using T1
ti
day:
Asterisk --x-- STP A ---x--- Switch1,2,3,4,5,6,7,8
STP B
Where Asterisk has voice CICs with all 8 switches, and all signaling
needs to be shared across a pair of signaling links, one with each STP.
Specially with E1s with all 8 switches can't fit on a single Asterisk bo
need. The
same layout works beautifully with libss7 (on a single system layout).
Marcelo Pacheco
M2J Communications - Brazil
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any free support for anything.
On 07/12/12 07:58, Abdul Basit wrote:
These are good options. chan_ss7 need to be more optimized. I can test
in my environment if you can share some test cases.
What is the plan of including this patch in main stream?
On Thu, Jul 12, 2012 at 2:34 PM, Marcelo
Removing the F on sent numbers is very simple.
Original code:
void isup_set_called(struct isup_call *c, const char *called, unsigned
char called_nai, const struct ss7 *ss7)
{
if (called && called[0]) {
if (ss7->switchtype == SS7_ITU)
snprintf(c->c
:
return '#';
case 0xB:
return '*';
case 0xC:
return 'C';
case 0xD:
return 'D';
case 0xE:
rds,
Marcelo
On 07/30/12 07:06, Kaloyan Kovachev wrote:
On Sun, 29 Jul 2012 18:18:48 -0300, Marcelo Pacheco
wrote:
Those who looked up the code might have noticed my "Original code"
didn't match libss7.
I changed digit handling code, so inside libss7 I always use 0...9 A...
No, THATS A BAD SS7 EXPERIENCE.
You must learn this BEFORE trying to mess around with ISUP.
ISUP is serious stuff, do it right.
If you haven't spent at least 2 full days studying ISUP and SS7, than
you're not even close to ready to mess around with ISUP in production.
On 08/02/12 14:38, David W
I got nothing to check.
I believe you didn't understood the discussion.
On 08/04/12 01:03, bipin singh wrote:
Hi,
Check your dialplan setting both side(switch and asterisk).
On Mon, Jul 30, 2012 at 10:49 PM, Marcelo Pacheco <mailto:marc...@m2j.com.br>> wrote:
That's
SS7 on Asterisk is very different from large TDM/NGN switches
On Asterisk the whole thing runs on a single process
On large switches, there are layers that talk to each other. Many design
features of SS7 are geared towards modeling failure modes that just
don't happen with Asterisk.
On 10/16/1
contact me off list.
Next feature is clustering, allowing for ss7 routes between the same opc
and dpc split between multiple asterisk instances.
--
Atenciosamente,
Marcelo Pacheco
M2J Comunicações e Informática
Fixo: (27)-8118 / (27)2233-2296
Vivo: (27)9964-5440
Claro: (62)9161-9047
MSN
have libss7 on one side and a TDM switch / STP with a proper MTP2
implementation, the other side will fault the link, but still, this is
very sloppy implementation. This would never pass a complete
certification test.
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Atenciosamente,
Marcelo Pacheco
M2J Comunicações e Informática
Fixo: (27)-8118 /
bss7 needs a lot of extra features to be proper, beginning with
retransmissions.
Marcelo Pacheco
On 12/02/12 23:03, Marcelo Pacheco wrote:
Since I developed the basic transport of MTP2 over UDP, I'm now able
to use iptables to cause MTP2 messages to be dropped and watch the
result, whe
ression is that libss7 was a pet project from Matthew Fredrickson.
Marcelo Pacheco
On 01/09/13 16:30, Ryan Crowder wrote:
>
> Is it just me or is LIBSS7 not available on asterisk.org anymore? Is
> it dead? Is chan_ss7 the preferred method no
ding charge indicator = 0. I'm using asterisk 1.8.18.0
>
> Thanks,
>
> Laura Maurizi
>
>
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Regards,
Marcelo Pacheco
M2J Comunicações e Informática
Mobile: 55(27)-8118
Landline: 55(27)2233-2296
MSN: marc...@macp.eti.br
Google Talk: marc...@enterchip.com.br
E-mail: ma
r myself.
But if you intend on doing really serious stuff with libss7, good luck,
it's not designed for that.
Just a sample of the now almost 20 issues I found, perhaps now I have
only 5 unfixed issues.
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Atenciosamente,
Marcelo Pacheco
M2J Comunicações e Informática
Fixo: (27)-8118 / (27
mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-ss7
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>
Per usual, read the fine manual. Wait, there's no manual !
Since you seem to have done your part and actually knows some ss7 and
isup, here comes a hint.
You created two or more linksets where you must have a single one.
libss7 don't have the ss7 routing feature.
In libss7 linkset concept is difer
ing the cic
>>> [1] Got RLC but we didn't send REL/RSC on CIC 10 PC 4097 reseting the cic
>>>
>>> Again, it's questionable, why this happened, but the second line seems
>>> to indicate some brokeness again.
>>>
>>> To explain: The
;> rejects RLC, which comes back as a response to the RSC which was just sent
>>>>> upon expiry of T17. And this appears again and again in the rhythm of T17,
>>>>> and the channel is not operational.
>>>>> ss7 show calls shows the following line for the misbehaving
There are no stupid questions, correct.
However most of us that started using the internet back when nettiquete
was the rule of the land, know that asking questions that have been
asked dozens of times before is rude.
RTFM was coined a long time ago and Google is your friend is as old as
Google's
stable
>> SS7, so I'm almost sure that they are not making signalling errors.
>>
>> With regards,
>> Pavel
>>
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>> asterisk-ss7 mailing
m GPL standards (customers get the
source), but I don't publish the source publicly.
--
Atenciosamente,
Marcelo Pacheco
M2J Comunicações e Informática
Fixo: (27)-8118 / (27)2233-2296
Vivo: (27)9964-5440
Claro: (27)9312-5319
MSN: marc...@macp.eti.br
E-mail: marc...@
Yes, most of those parameters are per CIC.
It should work like this:
ss7_calling_nai=A
group=1
channel => 1-16
ss7_calling_nat=B
group=2
channel => 17-31
On Thu, May 15, 2014 at 4:31 AM, Wasim Baig wrote:
> google for ss7_called_nai=dynamic
>
>
>
>
> On 15 May 2014 10:09, joseph mpora wrote:
>
Hello,
Is there a patch for an older version of asterisk, preferably asterisk 11
to run libss7-2.0.0 ?
Marcelo Pacheco
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To
I have my own extensive unpublished patch for Asterisk 1.8 that fixes lots
of SS7/ISUP issues for me.
I tried Asterisk 11 with unpatched sources and found lots of issues again.
I couldn't even get a stable environment with just all trunks aligned.
So I'm not surprised you're having problems.
No, my
gt; this should be published as libss7 2.0. Kaloyan did a lot of work on this
> and I really don't see _any_ issues now.
>
> --
> Michal Rybarik
>
>
> On 04/19/2016 07:27 PM, Marcelo Pacheco wrote:
>
>> I have my own extensive unpublished patch for Asterisk 1.
SS7-27 patches, but I debugged it
>> and Kaloyan fixed it then, fix is included in official sources now. All
>> this should be published as libss7 2.0. Kaloyan did a lot of work on this
>> and I really don't see _any_ issues now.
>>
>> --
>> Michal Rybarik
>
It doesn't work like that. I *strongly* suggest properly learning how SS7
works before trying to setup SS7. Your question makes it clear you haven't
done your homework.
Good luck.
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I detect a bug in the latest libss7 where for each group reset message, the
first CIC of each range doesn't get processed (staying pending), but every
other CIC does. Can you confirm that's what you see. For instance if its a
full E1, the first channel of the E1 stays pending. For an E1 with a
sign
g fixed if I could but I don't know where
> to start.
>
>
>
> On Mon, Feb 19, 2018 at 7:02 PM, Marcelo Pacheco
> wrote:
>
>> I detect a bug in the latest libss7 where for each group reset message,
>> the first CIC of each range doesn't get processed (staying pen
> > Anyone interested in fixing this for a bounty?
> >
> > FROM: asterisk-ss7-boun...@lists.digium.com
> > [mailto:asterisk-ss7-boun...@lists.digium.com] ON BEHALF OF Marcelo
> > Pacheco
> > SENT: Tuesday, February 20, 2018 16:16
> > TO: asterisk-ss7@lists
re' comment - the GRS is sent from the next
> case, so if you have the timer defined GRS will be sent every 5 minutes
> for ITU and every minute for ANSI with the default values
>
>
> На 2021-01-17 23:01, Marcelo Pacheco написа:
>
> > Here's one scenario this happ
Notice the big messages, a mere LSSU is 20 bytes. An ACK is 16 bytes.
FSN/BSN are 3 byte quantities.
[1] Link state change: INSERVICE -> IDLE
[1] Len = 20 [ 01 00 0b 02 00 00 00 14 00 ff ff ff 00 ff ff ff 00 00 00 09 ]
[1] >[2:0] LSSU SIOS FSN 16777215 BSN 16777215
[1]
[1] MTP3 T19 timer stopped P
In my first attempt to use chan_ss7, I configured my E1s the same way I
configured ISDN, except for I removed crc4 correctly.
So using dchan for SS7 signalling channels cause the error.
That resulted in lots of:
mtp.c: Excessive poll delay
So that's one more for the FAQ.
Regards,
Ma
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Marcelo Pacheco
Technolo
latform (Asterisk/Cisco as the Media Gateway + the
soft switch as the call distribution agent).
Here we use the ITU way of doing things, without indirect point code
routing and so far we only need ISUP, as LNP isn't quite here yet (should
be up and running in about 18 months).
--
Regards,
M
he current trunk bad ? I noticed core set verbose and core set debug
isn't working properly.
Any sugestions ?
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Marcelo Pacheco
Fale Voip Telecom-Brazil
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T
ng.
I got chan_ss7 working ok, but with frequent CRC/frame too short errors.
I'm using a Version 9 TDM405P, I can't ship it back to USA as it's in
production for the other 2 spans.
Marcelo
> On Nov 5, 2006, at 9:04 PM, Marcelo Pacheco wrote:
>> Most IAM and RLC result in
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> __
Line 11, add #include after #ifdef LINUX
Without this, compiling with -DLINUX fails
was:
#ifdef LINUX
#include
#endif
Change to:
#ifdef LINUX
#include
#include
#endif
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Regards
Marcelo Pacheco
Diretor de Tecnologia e Sistemas - FaleVOIP Telecom
Com: (27)2127-9791
Cel: (27)9945-3993
Fax
,
And ss7linktest moves forward.
I got the code with:
svn co http://svn.digium.com/svn/libss7/trunk libss7
Please let me know if I got code from the wrong place.
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Marcelo Pacheco
Diretor de Tecnologia e Sistemas - FaleVOIP Telecom
Com: (27)2127-9791
Cel: (27)9945-3993
Fax: (27)2127-9799
I belive chan_ss7 can do this, without a single A-Link per box, the
feature name is clustering.
However I have not used or tested this feature yet.
LES.NET (1996) INC. wrote:
> Hello.
>
> I'm curious if anyone has had asterisk-ss7 working (through whatever
> magic) across multiple boxes under a si
The easiest way to do this is to use libss7 + dahdi's native ss7
support, unless you want to look at all those repeated FISUs and LSSUs.
Also this way you don't need to perform HDLC decoding, each MTP2 message
comes as a single read/recv.
Satish Chandra wrote:
> Hi All,
>
> I am trying to develop
is to have complete CIC range based ISUP
masquerade. You can work around using more point codes, but that's
certainly sub optimal.
Marcelo Pacheco
Technical Director
M2J Communications
voip me wrote:
> Hi,
>
> I think its the best solution because handling complexity of
> distribute
Wrong. Asterisk will not function as a pure converter. All calls will
need to be switched, so if you intend on having 8 E1 coming from the
EWSD, you will need 16 E1 on Asterisk.
That is doable using 2x8 E1 PCI xpress cards, or a 4 PCI slot
motherboard and 4 quad PCI E1 cards. You'll need to disable
exclusively on interconnects.
Matt, what do you think ? Do you have a better suggestion than an Abort
message ?
Regards,
Marcelo Pacheco
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This can be done using 2 Asterisk instances.
Then you can set each one with its own OPC.
It might be possible with a single instance, not sure.
In order to run two instances of Asterisks on the same machine, you'll
need two sets of config files, each one needs to use diferent ports for
each VOIP se
libss7 can do it out of the box.
I currently have one Asterisk instance with one ethernet span talking to
another ethernet span (2 ethernet interfaces, conected to the same switch).
Nothing special, just set each 2 E1 groups with the settings you'd like.
Marcelo Pacheco wrote:
> This can
Dahdi groups is the feature you want.
It's well documented in chan_dahdi.conf
Antoine Megalla wrote:
> Hi,
>
> I did not find a way to do that in chan_ss7.
> However you can use the hunting_policy in the linkset definition to
> try achieve that in a way
>
> you can use
> hunting_policy => seq_
quot;)) {
Regards,
Marcelo Pacheco
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n, I'm all
ears.
Marcelo
Matthew Fredrickson wrote:
> Marcelo Pacheco wrote:
>
>> Hi Matt and asterisk-ss7 members,
>>
>> Looking at the dahdi-linux code, there seems to be an important feature
>> lacking on the dahdi mtp2 feature:
>>
>> Should the mtp
libss7 doesn't talk SCCP, and the Ericsson switch is trying to:
Network Indicator: 2 Priority: 0 User Part: SCCP (3)
Ask the mobile operator why they can't just use ISUP for everything !
resea...@businesstz.com wrote:
> Now i have another problem.. I have installed the new link to the Er
Obviously I miss comunicated. My point was asking the mobile operator
not to use SCCP, use only ISUP.
ISUP is for straight voice.
SCCP is for pretty much everything else, including SMS.
There's a proper MTP3 that Asterisk can reply to SCCP traffic that tells
the other switch it can't talk SCCP, ins
ctura (operating as an STP), Tropico RA and some others, without link
instability.
Anyhow, libss7 using dahdi mtp2 shouldn't suffer from this issue, as
well as chan_ss7, as both generate FISUs not stop correctly.
Regards,
Marcelo Pacheco
Kristian Nielsen wrote:
> "Gustavo Marsic
acts in that way, but it's very uncommon. As I
> tested a few years ago, EWSD in V11 (V13 and V15 works good) and 5ESS
> V14 can fail with that behavior.
>
> Regards,
>
> Gustavo
>
>
> On Jul 3, 2009, at 8:27 PM, Marcelo Pacheco wrote:
>
>
Used this with chan_dynamic of type loc (a virtual loopback span, with
100 channels) channels 1-100 talk to 101-200
networkindicator=national
echocancel=yes
echocancelwhenbridged=no
signalling=ss7
ss7type=itu
context=default
group=0
linkset=1
pointcode=1
adjpointcode=2
defaultdpc=2
cicbeginswith=
ves.
Your reply is the same as if I offered you a script to connect ports 1
and 4 of a 4 span card and you have a 2 span card, and you discard the
tip because you can't do basic math...
GOOD LUCK... YOU'LL NEED IT...
Vahan Yerkanian wrote:
> On 2/24/10 5:51 PM, Marcelo Pacheco wrot
Hi Enrico,
1 - Incoming/Outgoing channels have their differences
2 - Use core show channel , example: core show channel
SIP/278118-0144
3 - I currently don't have any TDM spans, so I won't even try to send
you an example output
4 - Also for every call there's an incoming and an outgoing ch
Mosbah,
If you're trying to test SS7 on the cheap, you can use TDMoE, that
requires no TDM hardware.
You can even have two Asterisks on the same machine.
Forget about FXS/FXO/BRI, that won't work ever.
TDMoE will simply emulate an E1/T1 trunk (actually it will do any size
trunk you want, I tried l
I had a quad E1 PCI setup with a PRI to the PSTN and a PRI to an AS5300,
the AS5300 was exactly handling analog modem calls.
Works ok, as long as the CPU is always lightly loaded.
Changing one E1 from PRI to SS7 should make no diference whatsoever if
you're using libss7, since libpri uses DAHDI and
I have tested chan_ss7 using TDMoE perfectly.
It was 2 years ago, there where issues using purely local TDMoE
configuration (without sending packets between two machines) but that
particular bug was fixed in DAHDI just a few months later.
ISDN works as well.
I created a 100 channel span for ss7 use
TDMoE generates 1000 interrupts/second just like any other DAHDI card.
Interrupts is not the issue. 1000 irqs/sec is doable with much older
processors.
What you need to worry about is echo canceling, transcoding, and other
issues.
Also, a single TDMoE span can be used for at least 120 channels. TDM
To clarify, I'm talking about TDMoE between two asterisks. If you want
to use TDMoE to use ethernet based E1 interfaces, you'll need to see how
that hardware works.
This channels per span stuff needs to be accepted from both sides.
Marcelo Pacheco wrote:
> TDMoE generates 1000 inte
What you're saying doesn't make a lot of sense.
We say OPC for the local point code and DPC for the destination point
code. It looks like someone switched the two.
It looks like you have one OPC and 4 DPC instead.
I've never seen any reason for multiple OPCs on a single switch except
for huge soft
On 04/26/11 18:18, Jean Louis David wrote:
Hi,
because E1 cards are quite expensive, and before connecting to PSTN,
i would like to test good working of asterisk + libss7 by looping back
2 E1 ports of the same PCI Card (for example : one TE410P). I will
connect two SIP phones for test purpos
Both ISUP/SS7 implementations commonly discussed in this list use an
E1/T1 card no differently from someone running ISDN over the card.
What matters is a stable driver, zero bit rate errors due to card
issues, and the availability of hardhdlc (hardware hdlc framer, depend
on the driver having th
Even a full SS7 STP is an routing/load balancing/plus simple ACL security.
STP is a level 3 functionality.
Proxy is level 4 or level 7 functionality.
Although some STPs come with extra funcionality that allows for
portability lookups and billing independent of the participating
switches, but cal
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