We were able to implement Operator Redirect via RLT to our Nortel DMS-100. It
was pretty much a hack but it worked for us. We completely changed how the FAR
message is sent to the switch and can call it from the dialplan.
--
_
-
You can also use Sangoma A104d cards.
I have purchased and used both digium and Sangoma cards.
Thank you.
On 11/9/11, bipin singh wrote:
> Hi,
> If you are new for asterisk then use only digium card . All
> digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you
> re
Hi,
If you are new for asterisk then use only digium card . All
digium pri (1 port , 2 port or 4 port ) card work with ss7 link . If you
required good voice quality
then use only echo cancellation digium card . We are using all card and
work fine with ss7 link .
On Tue, Nov 8, 2011 at
For SS7 you want a Digital T1/E1 card, which you can find here
http://store.digium.com/products.php?category_id=2
Which card you pick is simply a matter of how many E1/T1 you want to
support.
Torrey
On 8 November 2011 15:27, Rafael Machado wrote:
> Greetings to all the list.
> I make a purcha
I use TE410P for a year now, with 80-90 simultaneous calls, with no single
problem. I even forgot what whas the card, lol.
On Tue, Nov 8, 2011 at 6:27 PM, Rafael Machado wrote:
> Greetings to all the list.
> I make a purchase of a Digium card and the recommendation of friends would
> like to indi
Hi,
First check the caller id you received
NoOp(${CALLERID(all}).
and second issue use Set(callerid=XXX) or talk to telco and
set the default cli for ss7 link .
On Mon, Feb 14, 2011 at 10:05 AM, Trung Nguyen Dac wrote:
> Dear all.
>
> I have a trouble with making a
Hi José,
have you checked that the Telco clocking is provided on the correct timeslot
and that the SS7 signalling from the Telco is on the same timeslot as your
Asterisk system is expecting?
If you have an ISDN tester with audio monitor, you can listen for the
presence of SS7 signalling (sounds l
2011/1/25 José Pablo Méndez Soto
> Hi,
>
> We need assistance aligning 8 E1 ports (2x Digium Quad Span TE410P) with
> our TELCO.
>
> We are working on a relatively big SS7 deployment for some time now. We are
> using:
>
> asterisk-1.8.2.1
> asterisk-addons-1.6.2.
> dahdi-linux-complete-2.4.0+2.4
Hi,
Use asterisk-1.6.0 its working fine .
On Mon, Nov 29, 2010 at 7:04 PM, Timothy Smith wrote:
> Dear Users,
>
> I seeking help on with the asterisk+libss7. the call is successfully
> setup but no audio either end.
>
> I am using Asterisk SVN-branch-1.6.0-r265498, libss71.0.2,
> chan_da
problem and my CIC was not lined up. My T1-1 was their
>> > T1-5.
>> >
>> >
>> > Thanks,
>> > Dave
>> >
>> > -Original Message-
>> > From: asterisk-ss7-boun...@lists.digium.com
>> > [mailto:asterisk-ss7-boun...@li
And, I'm not really familliar with libss7 config, BUT..
Don't you have to declare 4 links (or groups) under 1 linkset in your
chan_dahdi.conf? in you current confi you have only 1 group.
try something like this:
linkset=1
group=1
sigchan=1
cicbeginswith=2
channels=2-31
group=2
sigchan=
cicbeginsw
> I had a similar problem and my CIC was not lined up. My T1-1 was their
> > T1-5.
> >
> >
> > Thanks,
> > Dave
> >
> > -Original Message-
> > From: asterisk-ss7-boun...@lists.digium.com
> > [mailto:asterisk-ss7-boun...@lists.digium.com
vember 29, 2010 9:57 AM
> To: asterisk-ss7@lists.digium.com
> Subject: Re: [asterisk-ss7] Help with SS7 (No Audio)
>
> Thank you Gentlemen for your responses.
>
> I have done the dahdi_monitor, its only TX that has some input (see
> sample output below). Thats for both outgoing and in
Behalf Of Timothy Smith
Sent: Monday, November 29, 2010 9:57 AM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] Help with SS7 (No Audio)
Thank you Gentlemen for your responses.
I have done the dahdi_monitor, its only TX that has some input (see
sample output below). Thats for both
Hi,
I havent seen any useful information while loading and unloading
chat_dahdi.so, however, I have attached below the output of
ss7linktest and a debug during a call.
By the way, what does it mean when RX is 0 (asterisk not receiving
anything from the telco?)
Thanks
Tim
ed channel
do you have any relevant logs on asterisk console.
set verbosity 3
unload chan_dahdi.so
then load chan_dahdi.so
you should see the ..cic expected on logs.
try to set that cic as cicbeginswith.
On Mon, Nov 29, 2010 at 7:56 PM, Timothy Smith wrote:
> Thank you Gentlemen for your
Thank you Gentlemen for your responses.
I have done the dahdi_monitor, its only TX that has some input (see
sample output below). Thats for both outgoing and incoming calls.
How can I verify the circuit mapping? My core engineer (telco company)
said that he is using the 1st channel for signalling
I had similar problem BUT with chan_ss7. it was because my CIC mapping was
wrong.
On Mon, Nov 29, 2010 at 5:34 PM, Timothy Smith wrote:
> Dear Users,
>
> I seeking help on with the asterisk+libss7. the call is successfully
> setup but no audio either end.
>
> I am using Asterisk SVN-branch-1.6.
Try sending a call via call file and see if you are getting both call legs.
callchannel.sh
#!/bin/bash
echo "Channel: DAHDI/$1/$2
Callerid: $2
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: ss7
Application: Echo" > /var/spool/asterisk/tmp/test.call
mv /var/spool/asterisk/tmp/test.call /var/sp
Any particularly reason why your CIC begins with 2? I think they usually
starts at 1. Check with the other side of the connection on CIC
configurations.
On Mon, Nov 29, 2010 at 8:57 AM, Abdul Basit wrote:
> Try sending a call via call file and see if you are getting both call legs.
>
> callchan
zaptel.conf:
loadzone = cn
defaultzone=cn
span=1,1,0,ccs,hdb3
bchan=1-31 # set this to 1-15,17-31 for E1
2009/10/17 zhubinyu
> Hi all:
> When i load chan_ss7.so,i get a error message like that,what is the
> problem!? STAMP-ASTFIN*CLI> module load chan_ss7.so
>
> STAMP-ASTFIN*CLI> [Oct
; Is it miss some lib like libc.so.6,or something else!?thanks
>
> zhubinyu
> 2009-10-16
>
> 发件人： Nahuel Greco
> 发送时间： 2009-10-15 21:23:18
> 收件人： asterisk-ss7
> 抄送：
> 主题： Re: [asterisk-ss7] help for chan_ss7.so module load er
lib like libc.so.6,or something else!?thanks
zhubinyu
2009-10-16
发件人： Nahuel Greco
发送时间： 2009-10-15 21:23:18
收件人： asterisk-ss7
抄送：
主题： Re: [asterisk-ss7] help for chan_ss7.so module load error!
In the target platform check the output of:
ldd /usr/lib/asterisk/modules/chan_ss7.so
Saludos,
Nahuel
In the target platform check the output of:
ldd /usr/lib/asterisk/modules/chan_ss7.so
Saludos,
Nahuel Greco.
On Thu, Oct 15, 2009 at 4:50 AM, zhubinyu wrote:
> Hi all:
> I had compiled chan_ss7 support for asterisk run on blackfin uclinux
> wiht blackfin uclinux toolchain.I get error l
Thanks Matt
Yes, I am familiar with C/C++ programming though not in a good shape at
the moment.
I will try my level best to get you online on MSN in your most convenient
time (our time diff is +8hrs )
Regards
Sam
___
--Bandwidth and Colocation Provid
resea...@businesstz.com wrote:
> Hi List
>
> I need to use ss7 optional parameter 'location number' on 911 (112) call
> centre implementation to determine the location of the calling part as
> passed by the mobile carrier
Hey Sam.
Sorry, I usually try to respond to most emails that nobody else h
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
"Satish Chandra" writes:
> I am trying to develop a Tapping device that can be used in a SS7 network. I
> am
> not able decide on a particular channel type which I should configure in
> zaptel.conf. The device needs to sit on a SS7 (T1/E1) link. As I will be
> duplicating signal using a tapping
0P card in asterisk box. I thought you were using
a TE410P as the converter.
Regards,
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: July 16, 2008 3:32 PM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] Help w
TE410P as the converter.
Regards,
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: July 16, 2008 3:32 PM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] Help with libss7 configuration
Hi Christopher,
Quoting
.
I use the crossover cable to connect the two ss7 linksets together.
Regards
Mark.
>
>
> Is this right ?
>
>
>
> Regards,
>
>
>
> Christopher
>
>
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark Wilkinson
> Sent: Ju
@lists.digium.com
Subject: Re: [asterisk-ss7] Help with libss7 configuration
Hi Christopher,
I'm not sure I understand your question.
What I have set up for testing SS7 is as follows :-
SS7 Linkset 1 comes out of TE410P span 1, via a T1/E1 crossover cable into
span 2 of the same TE410P t
Hi Christopher,
I'm not sure I understand your question.
What I have set up for testing SS7 is as follows :-
SS7 Linkset 1 comes out of TE410P span 1, via a T1/E1 crossover cable
into span 2 of the same TE410P to SS7 Linkset 2
Regards
Mark.
Christopher Srinivasa wrote:
How were you able
How were you able to go from two SS7 linksets into the TE410P E1 card to a
single ISDN after conversion on the crossover link?
Did you not have traffic congestion problems when going from 2 to 1 ?
Unless an SS7 linkset is equivalent to half an E1 ?
Anton:
I was mistaken. At last test sig mtp2 did NOT work with Sangoma cards.
The cards did work with sig = dchan. I'm going to delve further into this
and identify why not.
Apologies.
-wasim
On Thu, Jul 10, 2008 at 4:13 PM, Anton <[EMAIL PROTECTED]> wrote:
> Wasim,
>
> As I understand, mtp2
Wasim,
As I understand, mtp2 is hardwarely assisted sigchannel,
which functionality is implemented in DIGIUM cards to act
like dchan in PRI. I'd like just to confirm that we're
speaking about the same, and if so SANGOMA must have such a
support in their firmware too. As I know SANGOMA have
ha
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
Hi Anton,
I can only comment on it working with a Digium TE410P card
Sorry, I don't have any Sangoma cards.
Mark.
Anton wrote:
Is it only with DIGIUM cards or it works so with SANGOMA
CARDS too? Anyone tried?
On Thursday 10 July 2008 12:23, Mark Wilkinson wrote:
Hi Matthew,
I've just
Is it only with DIGIUM cards or it works so with SANGOMA
CARDS too? Anyone tried?
On Thursday 10 July 2008 12:23, Mark Wilkinson wrote:
> Hi Matthew,
>
> I've just tested the latest svn version with 'mtp2'
> instead of 'dchan' and my looped back linksets
> come up without any problems.
>
> Regard
Hi Matthew,
I've just tested the latest svn version with 'mtp2' instead of 'dchan'
and my looped back linksets
come up without any problems.
Regards
Mark Wilkinson.
Matthew Fredrickson wrote:
TCB wrote:
I could not get C7 to get up with mtp2= on dahdi, works fine with the
good ol' dchan
TCB wrote:
> Thanks Matt, Any work around on the ss7_calling_nai issue. I have a
> link that wont
> terminate the calls but inbound works fine.
>
I need to understand exactly what the issue is. Can you explain to me
what is wrong, and what needs to be fixed?
Thanks :-)
Matthew Fredrickson
>
Thanks Matt, Any work around on the ss7_calling_nai issue. I have a
link that wont
terminate the calls but inbound works fine.
On Wed, Jul 9, 2008 at 5:35 PM, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
> TCB wrote:
>> I could not get C7 to get up with mtp2= on dahdi, works fine with the
>> g
TCB wrote:
> I could not get C7 to get up with mtp2= on dahdi, works fine with the
> good ol' dchan. Feel free to try.
>
>
Just to let you guys know, I just fixed a bug that another developer
introduced into the mtp2 code in DAHDI last night. If you update your
DAHDI version to something new
Fantastic my linksets are now up !
Thanks for your help.
Is this a known problem with dahdi ?
Mark.
TCB wrote:
I could not get C7 to get up with mtp2= on dahdi, works fine with the
good ol' dchan. Feel free to try.
On Tue, Jul 8, 2008 at 9:03 AM, Mark Wilkinson
<[EMAIL PROTECTED]> wrote:
I could not get C7 to get up with mtp2= on dahdi, works fine with the
good ol' dchan. Feel free to try.
On Tue, Jul 8, 2008 at 9:03 AM, Mark Wilkinson
<[EMAIL PROTECTED]> wrote:
> Hello,
>I'm looking for some help configuring a system to test Asterisk's SS7
> setup.
>
> I want to have one syst
On Mon, 28 Aug 2006 17:33:47 -0300
Andre Luiz Martins Rodrigues <[EMAIL PROTECTED]> wrote:
> Aug 28 17:00:38 WARNING[4432] config.c: Missing interface entries for
> host 'gentoo1'.
you have to add a line like
if-1 => 123.123.123.123
to the host-section in ss7.conf, "if-1" is your network-device
Hi,
I am now testing the chan_ss7 v 0.8.2 and the SUS and
RES works great, Thanks to the Sifira guys :)
But I still have a problem with the ring back tone
when the caller is originating from the pstn side. I
can see that asterisk is sending ALERT_CALL in
progress but it seems that the Nortel
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