- quite often
it can be on TS1 when working with Siemens EWSD.
Regards
Darren
On Mon, 21 Nov 2022 at 13:44, Kaloyan Kovachev
wrote: Your link is misaligned. Check the configuration on the other
side - it
should provide the master clock according to your system.conf
На 2022-11-19 04:20, La
logs (before and after the
modification you suggested) the following message:
Huh?! chan_dahdi.c: [11] Huh?! Got FISU in link state 1. I've searched
the web but I couldn't find the meaning of this message.
El vie, 18 nov 2022 a las 8:40, Kaloyan Kovachev
() escribió:
Hello,
put sig
Hello,
put sigchan after the group definition:
group=1
cicbeginswith=1
channel => 1-15
cicbeginswith=17
channel => 17-31
sigchan=16
for now remove the slc line and check the configuration on the other
side: if it is configured to use different slc, then add it before
sigchan
На 2022-11-1
that message, those trunks stay pending
The workarounds are to restart dahdi when connectivity to the other
ISUP side is up
or use SS7 RESET GROUP linkset dpc cic range
Other switches will send GRS forever, until they get a response. See
ISUP timers T22 and T23.
Em ter., 31 de jul. de 2018 às 0
Hi,
please check if you have all the timers defined for both linksets more
specificaly ISUP timers T16 and T17
The proper functioning of libss7 depends on the timers and they need to
be defined for _each_ linkset. To avoid mistakes you may configure the
timers in a separate file (use ss7.ti
Hi,
can you try with Asterisk 13 instead and the same version of libss7.
This looks as mismatch by 1 bug, but for near 4 years (with 13.x) never
had similar problems, so it is most likely introduced at some later
point.
На 2018-02-20 16:24, Dovid Bender написа:
Asterisk version 15.1.4. Usin
Hi,
sigchan should be defined after the channel list and you should define
the ss7 timers, so your config should look something like:
... down to here all is the same
linkset = 1
pointcode = 5360
adjpointcode = 2500
defaultdpc = 2500
slc=0
cicbeginswith = 101
channel = 2-62
sigchan = 1
Hi,
You need to provide more info about your setup.
There are two SS7 options for Asterisk (chan_ss7 and lib_ss7) with
several versions each, so you need to tell us which version of Asterisk
and SS7 are in use, plus your config.
From your log, my guess is that you are using lib_ss7, but please
defined *after* the voice
channels
And this is also valid configuration for example:
cicbeginswitch=1
channel=2-62
sigchan=1
Then you have 61 voice channels numbered from 1 to 61
On Wed, Apr 20, 2016 at 6:33 PM, Kaloyan Kovachev
wrote:
The channel arrangement depends on how it is on the peer
Hi,
the problems you have posted 3 years ago were mostly for the stock
libss7 (trunk - partially 1.0, partially something else at that time).
Yes, it had many problems, but libss7 2.0, which have reached it's final
status and it's place in the official code base, was completed few
months later
On Wed, Apr 20, 2016 at 10:47 AM, Kaloyan Kovachev
wrote:
What kind of problems you had? Have you used timers in your config?
I am running several servers (with 2 or more trunks each) here for well
over 3 years (starting with Asterisk 11 and now 13) without any
problems and with peek loads that f
What kind of problems you had? Have you used timers in your config?
I am running several servers (with 2 or more trunks each) here for well
over 3 years (starting with Asterisk 11 and now 13) without any problems
and with peek loads that fill all 8 E1 on each server
Can't this be considered as
Hi,
as you are using Asterisk 13 with libss7 it is highly recommended to
enable all the timers not just mtp3_timer.t21 (from your other post).
To enable timers you need to explicitly define them or simply "#include
ss7.timers" (provided with samples) for each and every linkset
The sample con
e linkset and to add routing functionality, which is not so simple and
will require quite a lot of code changes.
Regards
M.Shirazi
On Mon, 3/21/16, Kaloyan Kovachev wrote:
Subject: Re: [asterisk-ss7] Asterisk with one signalling link but 2 opc
To: "
Hi,
Asterisk (with libss7) is a signaling end-point, so you can't have two
point-codes - you need an STP (which Asterisk is not) between the two
boxes in order to separate the two signaling channels.
However you do not need two point-codes for this setup, but simply to
define two groups of chan
Hi,
On 2016-03-19 02:10, Mehdi Shirazi wrote:
Hi
this post belongs to you?
http://lists.digium.com/pipermail/asterisk-dev/2012-August/056652.html
Yes, that is my post, but it's from a long time ago and I don't keep the
test setup anymore, but it is simple.
Your system.conf is correct:
dyna
Hi,
you are missing some definitions (networkindicator, ss7_called_nai,
ss7_calling_nai, ss7_*prefix) for the linksets and also to use mtp3/isup
timers (highly recommended) they should be explicitly defined
On 2016-03-17 10:15, Mehdi Shirazi wrote:
Hi
This is my system.conf :
dynamic=loc,1:0,
Hello,
there are lots of improvements since the version you are using. Please
test with libss7 v2 and Asterisk 13
On 2015-10-22 18:54, Joseph Jackson wrote:
Hi all,
Stats:
Asterisk 1.8.26.1
libss7 version: 1.0.2
DAHDI Version: 2.6.1 Echo Canceller: HWEC
I'm trying to get callerid present
Hello,
check https://issues.asterisk.org/jira/browse/SS7-27 i think the last
version of the asterisk part of the patch is
https://issues.asterisk.org/jira/secure/attachment/47909/Ast11_v7.diff
On 2015-05-17 06:46, Marcelo Pacheco wrote:
Hello,
Is there a patch for an older version of asteri
Hi,
patches are not accepted via mail-list.
You will need to create an issue on JIRA [1] and upload your patch there
after you sign a License Agreement
[1] https://issues.asterisk.org
On 2015-04-27 17:32, GOPI GANAPATHY wrote:
Dear all,
We are using the libss7 library along with asterisk fo
This question is more appropriate for the asterisk-ss7 list, but ...
Once you install libss7, you will need to rerun configure and recompile
Asterisk in order to use it
As you are using asterisk 11 keep in mind that it does not contain the
code for using the latest version of libss7 and you wil
libss7 does support multiple point codes and there is nothing special or
wrong with such configuration, but ...
When using Dial(DAHDI/g1/${EXTEN}):
If your linkset 1 is down, the calls go to the second linkset in your
group 1 - this is what you observe
If you have more than 60 calls they will s
Hi,
if you want to use timers - you need to explicitly define them or they
will not work at all.
Maybe at some point they will be enabled by default, but so far they are
not, in attempt to keep the behavior changes (and possible bugs) at
minimum if the new code is not used as it was a huge pa
No, I don't think that is the reason.
Can you please provide the output of both 'ss7 show cics 1' and 'ss7
show calls 1' at the same time when this happens.
Make sure you are using the latest libss7 - there was a bug, which was
fixed with https://reviewboard.asterisk.org/r/2150/diff/5-6/, whi
On Behalf Of Kaloyan
Kovachev
Sent: Tuesday, December 16, 2014 1:08 PM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] libss7 - two linksets with different DPCs
but the same CIC Number
Hi,
please show the output of "ss7 show cics 2"
On 2014-12-15 19:05, Lucian Raduc
7.conf.
I will try this config and get back with the results.
Regards,
Lucian
-Original Message-
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Kaloyan
Kovachev
Sent: Monday, December 15, 2014 4:08 PM
To: asterisk-ss7@lists.digium.c
assel
wrote:
libss7 supports this but chan_ss7 doesn't.
On 2014/12/15 11:19 AM, Kaloyan Kovachev wrote:
Hi,
it is absolutely normal to have such setup. Each linkset has it's own
CIC numbering
On 2014-12-15 01:06, Lucian Raducanu wrote:
Hi,
Is there any possibility to set up on th
Hi,
it is absolutely normal to have such setup. Each linkset has it's own
CIC numbering
On 2014-12-15 01:06, Lucian Raducanu wrote:
Hi,
Is there any possibility to set up on the same asterisk box 2 E1s with
different DPCs but the same CICs? To be more explicit, the first E1 has
CICs from 1
Hi,
in addition to FAXOPT(gateway) you may try to request transmission
medium from telco - see SS7_TMR or SS7_TMR_NUM, as it should be set on
the outgoing (dahdi) channel you need to set it on the SIP channel with
underscore (_SS7_TMR)
The possible options are defined in libss7.h and SS7_TMR_
Yes, there is some work done
(http://svnview.digium.com/svn/libss7/team/mattf/libss7-ss7cluster/) but
it's far from completed and may need some additional work to be merged
with the latest libss7
On 2014-03-14 12:44, Nitesh Bansal wrote:
Hello everybody,
I would like to set up SS7 point cod
Hi,
the config changes suggested from Michal (in both chan_dahdi.conf and
system.conf), should also fix your continuity check problems - just ask
the telco to repeat the test one you made them.
in system.conf you need the same changes:
#Sangoma A104 port 2 [slot:4 bus:4 span:2]
span=2,2,0,cc
And the 1st problem is - your link is looped back from the provider or
the equipment in between - you receive the message you have sent
yourself
On 2013-12-17 23:04, Attila Domjan wrote:
The 2nd problem: move the timer configuration lines before sigchan
definition.
2013/12/17 Vahan Yerkania
Hi,
On 2013-08-20 15:21, Frederic Van Espen wrote:
Hi,
There's contradicting information available online about redirecting
number and I'd like to have a clear answer on what works and what
doesn't.
I'm using libss7 trunk and asterisk 11.5.0 with the patches as
desribed in
https://issues.aster
tml
On Tue, Aug 13, 2013 at 6:38 PM, Akib Sayyed
wrote:
On Tue, Aug 13, 2013 at 6:23 PM, Kaloyan Kovachev
wrote:
On 2013-08-13 15:23, Akib Sayyed wrote:
Here is my config
; [span_1]
;signalling=pri_cpe
;switchtype=euroisdn
;pridialplan=national
;prilocaldialplan=national
;group=0
;co
chans from the same
span seems weird to me ... what is the topology of your connection with
the peer?
On Tue, Aug 13, 2013 at 5:51 PM, Kaloyan Kovachev
wrote:
On 2013-08-13 15:12, Akib Sayyed wrote:
Dear All
I am getting following message on my asterisk ss7 installation
ERROR[13077]: ch
On 2013-08-13 15:12, Akib Sayyed wrote:
Dear All
I am getting following message on my asterisk ss7 installation
ERROR[13077]: chan_dahdi.c:14445 dahdi_ss7_error: [2] !! Unable to
handle message of type 0x31 on CIC 3
WARNING[13077]: sig_ss7.c:424 ss7_find_cic_gripe: Linkset 2: SS7 GRS
requested
On 2013-08-13 14:12, Frederic Van Espen wrote:
adjpointcode=12248
sigchan=1
#include ss7.timers
sigchan=32
#include ss7.timers
adjpointcode=12198
sigchan=63
#include ss7.timers
sigchan=94
#include ss7.timers
you don't need multiple includes for the timers - just one per linkset
... the last
ode=1
defaultdpc=3
sigchan=1
sigchan=32
adjpointcode=2
defaultdpc=3
sigchan=63
sigchan=94
i.e. you should use a single linkset and if you want to separate some of
the channels - insert only 'context' and 'group' definitions before them
On 2013-08-13 12:12, Kaloyan Kovachev wr
Hi,
once you define sigchan - all channels (CICs) defined before that are
used for that link. When you define new channels after that - they will
belong to the next one.
Just move the block below at the end and you are done:
ss7type=itu
networkindicator=national
linkset=1
pointcode=4
adjpointc
original patch and the changes are quite big
already.
On 2013-08-13 10:47, Frederic Van Espen wrote:
On 08/12/2013 03:36 PM, Kaloyan Kovachev wrote:
No, all you need to do is described here
https://issues.asterisk.org/jira/browse/SS7-27?focusedCommentId=208465&
Hi,
On 2013-08-12 16:12, Frederic Van Espen wrote:
Hi,
On 08/09/2013 02:53 PM, Kaloyan Kovachev wrote:
I have tried to push the changes for inclusion to Asterisk 11, but
missed the deadline and now for Asterisk 12 too :(
The code published in the reviews is working flawlessly here for some
Hi,
On 2013-08-09 15:34, Frederic Van Espen wrote:
Hi,
I'm wondering what are the plans for the next libss7 release. 1.0.2
was released about 4 years ago, and the current code in trunk brings
major changes/fixes.
Are there any plans to make a real release soon or will it stay like
this for a w
Hi all,
i am happy to report, that i have (finally) put this version in
production with asterisk 11 and works very stable for a bit over a week
now.
I have uploaded the patch for asterisk 11 to JIRA [1]
The final changes made to libss7 (new trunk) can be downloaded from [2]
If you would like
Hi,
it's a (semi) blind guess, but ...
On 2013-07-25 14:59, Vallimamod ABDULLAH wrote:
Hello List,
I have a user asking me for a way to call a gsm phone in ss7 and reach
directly the voicemail without making the phone ring even it is
connected.
Does anybody know of a way to achieve this (tell t
Hi,
On 2013-06-27 07:48, Pavel Troller wrote:
Hi Kaloyan,
Hi all,
sorry for joining so late, but i am on holidays (by the end of the
week)
and rarely checking my mailbox. Thanks to bad weather i did that today
:)
Never mind, I'm happy you're here!
To the OP:
while reading the first posts
On 2013-07-01 10:14, Torrey Searle wrote:
You might also give this experimental ss7 branch a try. I have no
idea what its status is though
http://svn.digium.com/svn/libss7/team/mattf/libss7-ss7cluster/ [3]
That branch won't help as it is far from completed. There is some code
in isup_masq.
ffs from stock to your
branch.
On 06/26/13 09:05, Kaloyan Kovachev wrote:
Almost forgot. Please do not post patches (if any) in this list, but
attach them to the SS7-27 issue instead with proper license agreement,
so it can be included in Asterisk codebase
On 2013-06-26 14:57, Kaloyan Kovachev wrote
Almost forgot. Please do not post patches (if any) in this list, but
attach them to the SS7-27 issue instead with proper license agreement,
so it can be included in Asterisk codebase
On 2013-06-26 14:57, Kaloyan Kovachev wrote:
Hi all,
sorry for joining so late, but i am on holidays (by the
Hi all,
sorry for joining so late, but i am on holidays (by the end of the week)
and rarely checking my mailbox. Thanks to bad weather i did that today
:)
To the OP:
while reading the first posts i thought it is an old problem with
REL/RSC loop (persistent on start with ANSI signaling) which
Hi,
On Tue, 26 Mar 2013 14:05:16 +0100, Pavel Troller wrote:
> Hi!
> So, I'm getting familiar with this improved SS7 support. It seems
really
> improved compared to the standard one, but it still requires some
> modifications
> to be even better :-).
> 1) Overlap receiving support.
> It see
On Mon, 25 Mar 2013 11:27:51 +0100, Pavel Troller wrote:
> Hi!
> Ohh, so great! I was in the process of creating the patches... and
they
> are
> all available, and directly for V11! You really saved my day :-)!
> But many thanks to all the other people, who kindly responded and
> pointed m
On Thu, 07 Feb 2013 11:30:22 +0200, Kaloyan Kovachev
wrote:
> Hi,
> it seems i've messed that at some point, because it was something that
> was tested and working. I will have to resurrect my test setup in order
to
> fix it and update reviews 2150 and 2170, but unfortun
I am so glad to see you back Domjan
On Thu, 14 Feb 2013 14:39:37 +0100, Attila Domjan
wrote:
> Its an very old and fixed bug.
> We talked about it many times in this list.
>
--
_
-- Bandwidth and Colocation Provided by http://
Hi,
it seems i've messed that at some point, because it was something that
was tested and working. I will have to resurrect my test setup in order to
fix it and update reviews 2150 and 2170, but unfortunately it will not
happen in the next few weeks probably.
On Wed, 06 Feb 2013 04:13:03 +0100, M
Hi,
sorry for the delayed replay (been busy with hardware probs)
I hope you are not using ANSI, as it does not support overlap dialing. You
should also have defined (uncommented) the ISUP T10 timer
The function responsible for the release is ss7_match_extension() and it
sends it in case no extens
Hi,
you have missed an important piece of information as libss7 and asterisk
versions. Is this with the latest changes made ( see
https://reviewboard.asterisk.org/r/1676/ and issue SS7-27)?
On Wed, 10 Oct 2012 02:19:10 +0200, Gregory Massel
wrote:
> I've picked up a bug in libss7.
> The real
there, and
> Mobicents is the slave (TE_CLOCK = NORMAL in wanpipe1.conf) and so
> span=1,1,0,ccs,hdb3,crc4.
>
> So the clock comes from the Asterisk side.
>
>
> On Mon, Oct 1, 2012 at 11:46 AM, Kaloyan Kovachev
> wrote:
>
>> Who is providing the clock in your set
Who is providing the clock in your setup?
span=1,1,0,ccs,hdb3,crc4 - on Asterisk machine means it expects the clock
from remote, but if Mobicent does not provide clock you need:
span=1,0,0,ccs,hdb3,crc4
On Mon, 1 Oct 2012 11:29:13 +0200, Victor Neiman
wrote:
> HI Mitul,
>
> thanks for the sugg
Hi,
i think that mtp2=1 works only with Digium cards and you need to use
dchan=1 instead
On Mon, 1 Oct 2012 10:44:41 +0200, Torrey Searle
wrote:
> With wanpipe cards, I believe you need to also configure the timing
> information into the wanpipe config files too.
> (/etc/wanpipe/wanpipe*.conf)
>
Hi,
unfortunately in your version there are no much debugging tools for ss7.
ss7 show channels|calls|cics would be helpful here, but not available. My
guess is that by some reason the CICs get blocked and are never unblocked,
as you receive incomming calls ... In sig_ss7.c for case ISUP_EVENT_IAM
Hello and thank you for testing it
for libss7 you should get trunk, but also apply the patch from
https://issues.asterisk.org/jira/secure/attachment/44345/SS7-27_libss7_trunk2_v1.diff
for asterisk just download trunk and apply the patch from
https://reviewboard.asterisk.org/r/1676/diff/raw/
Hi,
please ignore my message, as i have found them documented in
team/mattf/bug13495 ... now it makes sense.
On Thu, 02 Aug 2012 12:20:45 +0300, Kaloyan Kovachev
wrote:
> Hello,
> i have ported Domjan's changes to trunk
> (https://reviewboard.asterisk.org/r/165
Hi,
On Thu, 02 Aug 2012 12:19:06 +0200, David Wilson
wrote:
>
> */etc/asterisk/chan_dahdi.conf*:
> [trunkgroups]
>
> [channels]
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforwar
Hello,
i have ported Domjan's changes to trunk
(https://reviewboard.asterisk.org/r/1653/ and
https://reviewboard.asterisk.org/r/1676/), but i have some problems
documenting the new config options: ss7_use_echocontrol and sls_shift
Will appreciate if someone is using them or otherwise can shed som
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...F
> digits. #->A and *->B translation happens early coming in, and on th
On Thu, 26 Jul 2012 12:54:58 -0300, Gustavo Mársico
wrote:
> I faced a similar issue recently and was related to the Screening
> Indicator. Was solved adding before Dial the following line:
>
> Set(CALLERPRES()=allowed) or Set(CALLERPRES()= prohib)
>
> In both cases the important was the 2 bits
Hi,
make sure your switchtype is properly configured.
ITU requires the number to be terminated with '#' which is 0xF
ANSI does not want/like the '#', so probably you have 'ss7type=itu' while
the other end is ANSI
On Thu, 26 Jul 2012 05:59:43 -0700 (PDT), Marcus Vinicius
wrote:
> Hi,
>
>
> I
om
>> [mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Torrey
Searle
>> Sent: Monday, April 02, 2012 10:01
>> To: asterisk-ss7@lists.digium.com
>> Subject: Re: [asterisk-ss7] Variable if call is national or not
>>
>> If you set the
>
In case ss7linktest is unable to get the link up asterisk configuration
changes will not help and there is no reason to experiment with that.
I think the problem is with the connection itself ... most likely a cable
at your side or telco side ... how is the link connected to the telco?
On Wed, 30
BSN: 127 BIB 1
> <[0] LSSU SIE
>
> Link state change: ALIGNED -> PROVING
> Len = 4 [ ff ff 01 03 ]
> FSN: 127 FIB 1
> BSN: 127 BIB 1
> <[0] LSSU SIOS
>
> Link state change: PROVING -> IDLE
> Link state change: IDLE -> NOTALIGNED
> Len = 4 [ ff ff 01 00 ]
&
Hi,
On Thu, 24 May 2012 06:47:28 -0700 (PDT), Marcus Vinicius
wrote:
> Hi,
>
> I can't alignment a libss7 Asterisk with a TDM switch (Huawei/UMG)
because
> the link doesn't pass the "PROVING" state.
>
> All configurations are correct according from telco, but I always get
the
> same logs:
ch
On Thu, 29 Mar 2012 09:59:03 +0200, "Dovid B" wrote:
> Hi,
>
>
>
> Is there some sort of variable that will let me know if the calling
party's
> number is in national format or not?
>
There is no such variable set by default, but you may:
1. use ss7 debug and look at the incoming message fo
te change: IDLE
-> NOTALIGNED") it is at provider's end or cabling
if it is missing at your end - run again dahdi_cfg without -t option or
even try to restart the machine
> --
> regards,
>
> abdul basit
>
>
>
> On Tue, Mar 27, 2012 at 7:27 PM, Kaloyan Kovache
is weird.
last idea ... if you replace one of the old sigchans (e.g. 109) with 140
does it work?
>
> --
> regards,
>
> abdul basit
>
>
>
>
> On Tue, Mar 27, 2012 at 6:01 PM, Kaloyan Kovachev
> wrote:
>
>> On Tue, 27 Mar 2012 17:35:34 +0500, Abdul
gt; mtp2=171
> echocanceller=mg2,156-170,172-186
>
> # Span 7: TE4/1/3 "T4XXP (PCI) Card 1 Span 3" HDB3/CCS
> span=7,7,0,ccs,hdb3
> bchan=187-201,203-217
> mtp2=202
> echocanceller=mg2,187-201,203-217
>
> # Span 8: TE4/1/4 "T4XXP (PCI) Card 1 Span 4"
16:46:05] Resetting CICs 209 to 223
> [Mar 21 16:46:05] Resetting CICs 225 to 239
> [Mar 21 16:46:05] Resetting CICs 241 to 255
> [Mar 21 16:46:06] MTP2 link up (SLC 3)
> [Mar 21 16:46:06] MTP2 link up (SLC 2)
>
> Checking with my provider. Will update after coordination.
yan,
>
> Thank you for suggestion.
> I will update and recompile asterisk. Will get back with results.
>
> --
> regards,
>
> Abdul Basit
>
>
>
> On Sat, Mar 17, 2012 at 12:47 PM, Kaloyan Kovachev
> wrote:
>
>> On Fri, 16 Mar 2012 23:25:40 +0500, Abd
On Fri, 16 Mar 2012 23:25:40 +0500, Abdul Basit
wrote:
> This is production machine. I found it as stable one and working fine
with
> me.
> Moreover, I tested with latest asterisk 1.6.2.23 but this is not solving
> slc problem.
>
> If someone has better idea/suggestion thanks in advance.
There a
th 1.8
In production i am (still) using 1.6.1 with the old version of the
patches.
>
>
> thanks
>
>
> --
> Marcus
>
>
>
>
>
> De: Kaloyan Kovachev
> Para: Marcus Vinicius
> Cc: asterisk-ss7@lists.digium.com
> Enviadas: Terça-feira, 6 de Março de 2012 10:
On Tue, 6 Mar 2012 05:05:26 -0800 (PST), Marcus Vinicius
wrote:
>
> Can you tell me if I having any concept mistake or if my configuration
is
> wrong?
>
your configuration seems correct and i have similar setup working with the
difference that i have only 2 not 3 E1 and i manually specify th
Hi,
On Mon, 5 Mar 2012 19:48:15 +0100, Torrey Searle
wrote:
> Do you have other links defined?
>
> Usually, libss7 should start counting from 0 for each signaling link
> defined in a linkset.
Yes this is true - the SLC starts from 0 and count up
>
> On 5 March 2012 19:33, Chris Twombly wrote
Hi,
move your sigchan definitions at the end - see below
On Mon, 5 Mar 2012 12:35:56 -0800 (PST), Marcus Vinicius
wrote:
> Hello,
>
> I'm trying to setup a cenario with 2 sigchan in a linkset. Each sigchan
is
> on an E1 link.
>
> I have 3 E1 links:
>
> span=1,0,0,ccs,hdb3,crc4
> bchan=1-31
Hi,
On Mon, 27 Feb 2012 04:54:39 -0800 (PST), Marcus Vinicius
wrote:
> Hi,
>
>
> Is there any way to setup the ss7 calling party category an outgoing
> call?
>
> I tried to setup the variable _SS7_CALLING_PARTY_CATEGORY, but it
doesn't
> work.
>
> I saw this topic from 2009:
>
> http://ww
Hi,
On Wed, 15 Feb 2012 12:53:15 +0530, Bharat Lalcheta
wrote:
> No i m not getting BAD FCS HDLC abort. In dahdi i m getting following 2
> messages.
>
> REL on channel (CIC 113) without owner! This message i m getting on
first
> server too.
>
> But below message came on the problematic server
On Sat, 21 Jan 2012 10:51:53 +0530, bipin singh
wrote:
> Hi,
> Provide the bug log ...
>
it is available in https://issues.asterisk.org/jira/browse/SS7-27
>
> On Fri, Jan 20, 2012 at 4:32 PM, Kaloyan Kovachev
> wrote:
>
>> On Mon, 16 Jan 2012 11:05
On Mon, 16 Jan 2012 11:05:04 +0200, Kaloyan Kovachev
wrote:
>>> On 01/04/2012 07:49 AM, Kaloyan Kovachev wrote:
>>>> Hello list,
>>>> most of us are using Domjan's version with Asterisk 1.6.x.x for
>> months
>>>> (over an year). Cur
>> On 01/04/2012 07:49 AM, Kaloyan Kovachev wrote:
>>> Hello list,
>>> most of us are using Domjan's version with Asterisk 1.6.x.x for
> months
>>> (over an year). Current Asterisk trunk does not even compile with
> libss7
>>> trunk.
>
On Wed, 04 Jan 2012 08:20:07 -0600, "Kevin P. Fleming"
wrote:
> On 01/04/2012 07:49 AM, Kaloyan Kovachev wrote:
>> Hello list,
>> most of us are using Domjan's version with Asterisk 1.6.x.x for
months
>> (over an year). Current Asterisk trunk does not even c
Hello list,
most of us are using Domjan's version with Asterisk 1.6.x.x for months
(over an year). Current Asterisk trunk does not even compile with libss7
trunk.
So i have decided to try to update the patches from Mantis ID 13495 to
current trunks so we would finally have (hopefully) a stable an
Hi,
you will need to make a patch for that as currently it gets a fixed value
- check isup.c for 'transmission_medium_reqs'
Also there is a bug as 3.1 kHz is marked to be 4 instead of 3
4 and 5 are unused
6 is also spare (like 1) and not 64 kbit/s preferred while the rest are
not used according
Hi,
there is no such thing like N with SS7 - 0123456789ABCD*# are the
possible options with '*' = 'e' and '#' = 'f'
On Fri, 11 Nov 2011 11:43:54 -0300, "Germán I. Paul"
wrote:
> Hi.
> Can someone of you tellme how can I add an N at the end of the
dialed
> extension using libss7?
> I`ve t
Hi,
check http://www.voip-info.org/wiki/view/Asterisk+libss7
I guess you meant 100k subscribers and not 100k simultaneous calls, so it
is doable with Asterisk and much less hardware then - each call will land
at Asterisk for just few seconds, so for the correct number of PRI cards
you need to che
Hi,
the configuration below should be OK except for:
> cicbeginswith=1
> networkindicator=national
> sigchan=108
> channel=94-107,110-124
should probably became
cicbeginswith=1
networkindicator=national
channel=94-107
cicbeginswith=17
channel=110-124
sigchan=108
so you can skip CIC 16 too
and
On Fri, 11 Feb 2011 11:04:49 +0100, mosbah abdelkader
wrote:
> Hi,
>
>
> Thank you for your help.
>
>
> For the timer, the doc mentions that a DAHDI card must be used What
to
> use for timer if there is no card? Or, the timer does not matter if no
> hardware is available?
>
You may use d
Hi,
On Fri, 11 Feb 2011 09:28:20 +0100, mosbah abdelkader
wrote:
> Hi,
>
>
>
> Is it possible to configure TDMoE for Asterisk and DAHDI without using a
> DAHDI hardware.
>
>
>
> If yes, please give some explanation.
>
yes it is possible.
In dahdi/system.conf you configure a dynamic span
On Thu, 10 Feb 2011 10:38:09 -0600, "Kevin P. Fleming"
wrote:
> On 02/10/2011 10:23 AM, mosbah abdelkader wrote:
>> Ok, I find it used in Asterisk 1.8.
>>
>>
>> My question remains for other versions: 1.2, 1.4 and 1.6.
>
> Those versions of Asterisk do not have SS7 support.
1.2 and 1.4 do not ha
n&revision=230
On Tue, 21 Sep 2010 14:54:51 +0200, Johann Steinwendtner wrote
> On 2010-09-02 14:49, Kaloyan Kovachev wrote:
> > check if your chan_dahdi.c supports setting slc at all. It should be in
> > function process_dahdi just after ss7type
> >
> > On Thu, 2 Sep
check if your chan_dahdi.c supports setting slc at all. It should be in
function process_dahdi just after ss7type
On Thu, 2 Sep 2010 07:37:05 -0500, Stephan Ellis
wrote:
> I am not sure about the difference of what Bipin sent, except for the
> order
> and apparent removal of directives. I tried
Hi,
the last message is SUS, so the call is suspended and if the called
person on the PSTN side pickups the phone again the conversation will
continue - there is nothing wrong with Asterisk, but caused from the setup
at the PSTN side. For most operators you will probably get release of the
call af
If you get audio after you press some DTMF - take a look at:
http://lists.digium.com/pipermail/asterisk-ss7/2010-April/003613.html
On Sun, 11 Jul 2010 12:57:57 +0500, Wasim Baig
wrote:
> On Sun, Jul 11, 2010 at 12:45, Basil Hweij
wrote:
>
>> Dear all,
>>
>>
>>
>> I setup libss7 1.0.2 with aste
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