google for ss7_called_nai=dynamic
On 15 May 2014 10:09, joseph mpora wrote:
> Hullo,
>
> We have 1 E1 from a telco that they would like us to use for both
> local and international traffic.
>
> I currently have ss7_calling_nai set to national and this works fine
> for local calls. Now they wo
kamran:
the issue is usually cic to dahdi channel mapping ...
dahdi_monitor is not the same as app_record
start dahdi_monitor 8 -v from linux shell
then make a call on cic 8
once call is answered, say something and you should see a visual
representation
same thing for the return media leg
if
Use dahdi_monitor to verify the CIC has media on it ...
Usage: dahdi_monitor [-v[v]] [-m] [-o] [-l limit] [-f FILE |
-s FILE | -r FILE1 -t FILE2] [-F FILE | -S FILE | -R FILE1 -T FILE2]
Options:
-v: Visual mode. Implies -m.
-vv: Visual/Verbose mode. Implies -m.
-l LIMIT:
On Mar 3, 2014 11:28 PM, "Miguel Molina"
wrote:
>
I'm afraid using only voice channels on second server won't work, because
the scope of the signalling channel in the first server will only be the
voice channels of the links connected to it.
Sorry, but this is incorrect. The scope of the sig lin
ChanSS7 comes with sample clustering conf.
Or you can swap one 4E1 card with one of the new Sangoma 16E1 card!
--
wasim h baig | +923008508070 | peace be upon you
On Jun 29, 2012 8:04 PM, "Camilo Echeverry"
wrote:
> Hi.
>
> Currently I have 12 E1 (3 x 4port Digium cards) with SS7, my hardw
; The SCCP global title: translation-type, nature-of-address,
numbering-plan, address
globaltitle => 0x00, 0x04, 0x01, 4546931411
ssn => 7
route => 929820405471:ra_geb, 929820367598:ra_geb, 929820706441:ra_geb, :ra_geb
-wasim
On Tue, Feb 7, 2012 at 10:07, Salman Minhas
wrote:
> Hello,
>
> I want
wow, that is excellent news Mr Mueller! takes me straight back years and
years ...
so, now we have chan_ss7, libss7 and ss7box as free, opensource options
... i wonder if someday we'll see libISUP as well?
the best place for distribution would probably be github ... let me know if
we can be of as
mike,
he means Custom Ring Back Tone, where a subscriber can have a db lookup
done for a song of their choice and played back to a calling party, instead
of the standard ring-ring
this is generally an INAP or CAMEL function, none of the existing SS7
options have it for asterisk in FOSS, afaik
10
AFAIK, asterisk-ss7 has had discussions on all SS7 options for asterisk
including libISUP, lib_ss7, chan_ss7, ss7box, SMG etc ...
Has the list policy changed to be solely lib_ss7?
-wasim
On Fri, Sep 2, 2011 at 14:36, wrote:
> Raul,
>
> ** **
>
> You are on the wrong channel with this quest
You may be better off using chan_ss7 2.0, with clustering and mtp3d and put
8-16 E1 per box.
Both chan_ss7 and libss7 can have multiple signalling channels. Standard
deployment is to use atleast 2 channels for redundancy, preferably on
different E1, even more preferable on different asterisk boxes
An E1 is an E1, irregardless of whether you run a PRI on it, or SS7 (these
are the signalling suites)
The end to end connectivity of an SS7 E1 should be the same as that of an
ISDN PRI E1.
- wasim
p.s. they aren't modems, no modulation/demodulation happens ... more likely
line drivers or amplifi
On Tue, Jan 11, 2011 at 22:57, Benaiad wrote:
>
> On Jan 10, 2011 9:59 AM, "Wasim Baig" wrote:
> >
> > On Sun, Jan 9, 2011 at 23:18, Benaiad wrote:
> >>
> >> Hi,
> >>
> >> ;First of all, I'm newbie in this field, so please
Jorge:
try second cic starting at 33, not at 32, 3rd at 64 etc ...
skip 1 channel at each span cic definition
-wasim
On Tue, Jan 11, 2011 at 23:04, Jorge Antillon wrote:
> Hi,
>
> Single DPC, single signalling channel, although we should have a second
> signalling channel available soon.
>
>
On Sun, Jan 9, 2011 at 23:18, Benaiad wrote:
> Hi,
>
> ;First of all, I'm newbie in this field, so please forgive
> my unprofessional explanation.
>
> I'm trying to connect to a mobile operator with chan_ss7 using sangoma
> card.
> the telecom regulator initiated an SPC code for us
> Now, the op
On Sun, Jul 11, 2010 at 12:45, Basil Hweij wrote:
> Dear all,
>
>
>
> I setup libss7 1.0.2 with asterisk 1.6.2.8 and link set is up and the call
> get answered but they hear nothing.
>
do you hear anything? or neither party does?
probably a cic mismatch
first check dahdi_monitor on the cic you
On Tue, May 25, 2010 at 11:18, bipin singh wrote:
>
> hi use libss7
>
> On Mon, May 24, 2010 at 6:17 AM, Geoffrey Yeoh wrote:
>
>> I have a project in which I need to do the following - Install two
>> trixbox server with Sangoma A101 card in each server and connect them to two
>> E1 ports on a Qu
On Mon, May 17, 2010 at 12:40, Arafath-uz-zaman khan
wrote:
> Hello all
>
> I was trying to configure total 32 E1( 16 E1 per server) using sangoma
> card. I got following msg when i am trying to run mtp3d " *config.c:707
> load_config_link Too many links defined for linkset 'MSC1' for link 'l17'
>
On Mon, Feb 8, 2010 at 3:58 AM, vallimamod abdullah <
vallimamod.abdul...@imtelecom.fr> wrote:
> Thank you very much for your hint: I needed to disable all hardware
> stuff (echo cancellation, dtmf and fax detection.)
> I also needed to disable SLTM with sltm => no in ss7.conf
>
> Now my link is i
ested config files:
>
> - wanpipe1.conf: http://pastie.org/813689
> - dahdi/system.conf: http://pastie.org/813703
> - ss7.conf: http://pastie.org/813698
> - output of wanpipemon: http://pastie.org/813692
>
> Thanks for your help.
> - vma
> .
>
> On Sunday7Feb, 2010, at
It seems as if you don't have mtp2 on the signalling link
please pastebin your /etc/wanpipe/wanpipe1.conf, /etc/dahdi/system.conf and
/etc/asterisk/ss7.conf
also, the output of wanpipemon -c Ta -i w1g1
assuming your siglink is on wanpipe1
-wasim
On Sun, Feb 7, 2010 at 11:33 PM, vallimamod abdul
On Fri, Feb 5, 2010 at 10:53 PM, ABBAS SHAKEEL
wrote:
> Please some one shed some light on it...
>
repetitive emails will generally not get you an answer ...
On Thu, Feb 4, 2010 at 4:03 PM, ABBAS SHAKEEL
wrote:
>
>> Hello All,
>> Please let me know Answers to the following questions .
>>
>> 1. W
rajesh:
use dahdi_monitor to see if the voice is actually going out on the
particular channel
or one above or below it, as its probably just a cic mismatch
-wasim
On Fri, Sep 18, 2009 at 12:11 PM, Rajesh Mahajan
wrote:
> Hi All.
>
> We are using Sangoma A104u Quad Card for SS7.
>
> Incoming ca
On Wed, Jun 10, 2009 at 12:18 AM, Krzysztof Drewicz <
krzysztofdrew...@gmail.com> wrote:
> As i said, im very new in SS7. Anybody knows the reason of the
>> limitation to 7 pris?. In other systems, we have 14 pris (not SS7) working
>> in a system that is exactly as this one with no problems. An
On Sat, Apr 18, 2009 at 5:26 PM, Kashif Ali wrote:
i wan to configure digium te412p card as ss7 interface. i installed
> libss7,dahdi and asterisk. Do i need to install lip pri also to work on ss7?
nope
--
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 |
peace be upon
On Mon, Oct 6, 2008 at 10:06 PM, Elkhomeini <[EMAIL PROTECTED]> wrote:
> hello,
> realy I don't know if I can put my queastion here or not ,
>
the correct mailing list would be asterisk-users
this one is for asterisk and ss7 related queries
>
> I would like to make asterisk call center , I ha
On Tue, Sep 2, 2008 at 8:26 PM, Virmones Pereira <[EMAIL PROTECTED]>wrote:
> Regards for everybody
>
> Joseph your solution is very interesting, but I think I will have another
> two problems the first trouble is with billing, Because if I answer the call
> it will be charged right?
> And the seco
Pawel at halo-kwadarat, has also done an analysis for SMG on 16E1, read his
analysis and thorough testing methodology at
http://wookieboo.halokwadrat.pl/lang-en/component/content/article/17-ss7boxsmg/29-ss7-smg-performance
Although this is SMG specific, it does show workable solutions of 16E1 per
On Wed, Jul 16, 2008 at 11:40 PM, Mitul Limbani <[EMAIL PROTECTED]> wrote:
>
> I think the Sangoma A108 would do the trick to get max E1s down to a
> single machine, but i also think that you may need to have 3U-4U kinda
> Chasis to fit in those many cards (1U may not work)
I've had good luck wi
know SANGOMA have
> hardware assistance in their card for the own SS7 solution,
> but will this work for Asterisk SS7 - is a question.
>
> On Thursday 10 July 2008 15:10, Wasim Baig wrote:
> > 2008/7/10 Mark Wilkinson <[EMAIL PROTECTED]>:
> > > Hi Anton,
> > >
2008/7/10 Mark Wilkinson <[EMAIL PROTECTED]>:
> Hi Anton,
>I can only comment on it working with a Digium TE410P card
>
> Sorry, I don't have any Sangoma cards.
>
I can confirm it works with Sangoma cards as well as ofcourse, Digium cards.
If you are using Sangoma cards, make sure you disabl
On Tue, Jul 8, 2008 at 7:58 AM, Rony Ron <[EMAIL PROTECTED]> wrote:
> - subscribers' call controlling
>
> - automatic call generation to test new services
>
> - SMS to voice & Voice to SMS
>
> - Extra VM solution
>
> - Closed User Group
>
most of the above should be fairly straight forward to impl
On Mon, Jul 7, 2008 at 11:11 PM, Mitul Limbani <[EMAIL PROTECTED]> wrote:
> Rony,
>
> Please throw some more light on the topic :)
> Mebbe we could have something already done out there...
>
> Quoting Rony Ron <[EMAIL PROTECTED]>:
>
> > Hello there,
> > anyone experienced in integrating our so nic
On 7/28/07, Fulvio Picecchi <[EMAIL PROTECTED]> wrote:
>
> Hi, how many links can a single box (with a single singaling link) handle?
> Is there any limit? Has anyone pushed chan_ss7 to the hardware limits (cpu
> usage, ram, etc.)?
fulvio:
the theoretical max for a single 64kbps signaling link i
On 6/6/07, Sam Njenga <[EMAIL PROTECTED]> wrote:
Has anyone had the asterisk ss7 type approved in any country by their
local Communications Authority ?
There are two things here.
One Type Approval needed is for hardware level certification, like FCC or
ETSI. These are available for the more
On 4/10/07, Vazir <[EMAIL PROTECTED]> wrote:
Matthew,
I can give you an ssh access to a pc and will reproduce the
case in a few minutes
Or tell me how to get full backtrace of the treads.
http://www.voip-info.org/wiki/view/Asterisk+debugging
Backtracing a core dump file in /tmp
1. start
As umar said, the simplest solution to this is to set a dummy callerid. You
can check for existence or validity of callerid in the dialplan, else
set(${CALLERID(number)=2342342) etc or whatever your switch will accept.
-wasim with top posting madness
On 3/16/07, Vernier Umali <[EMAIL PROTECTED]>
On 11/18/06, M Jamshed <[EMAIL PROTECTED]> wrote:
I'm having a problem with the PRI which I'm using for Dial Outs at
my client side. Its like in between dial outs PRI's are getting down and up.
When i have discussed with my clients, they are telling like its my end
problem since their part
apc is for the "next hop router" on an ss7 network, its what your sp talks to ...umar, it is my recommendation that you read up on ss7 on google ...i'm sure you'll agree that instead of the developers spending time with basic info questions
we need them to be concentrating on MUCH more crtiical thi
On 10/10/06, M Jamshed <[EMAIL PROTECTED]> wrote:
how ll i use that
On 10/10/06, Florian Overkamp <[EMAIL PROTECTED]
> wrote:Did you use Playback with the 'noanswer' option ?
jamshed:i and others have told you three or four times now, to use noanswerits an option to the Playback commandshow app
On 10/9/06, M Jamshed <[EMAIL PROTECTED]> wrote:
Dear,
I' m having a big problem now. please sort it out now...
The matter is that.. if all our services are down then i need to
play a voice prompt which should not be charged for the subscriber.. is
there any solution for this,,,
please tell me ou
On 6/22/06, Tawanda Charles Bwanya <[EMAIL PROTECTED]> wrote:
Anyone done a successful implementation of Asteriska and SS7.Please advise on how its working for you .there are a number of successful deployments with asterisk and the three
flavours of ss7 you currently get, cosini, xygnada and sifira
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