Hi,
Forget my previous email the timer configuration is correct.
2013.12.17. 22:04 ezt írta ("Attila Domjan" ):
> The 2nd problem: move the timer configuration lines before sigchan
> definition.
>
>
> 2013/12/17 Vahan Yerkanian
>
>> On Dec 17, 2013, at 8:24 PM,
The 2nd problem: move the timer configuration lines before sigchan
definition.
2013/12/17 Vahan Yerkanian
> On Dec 17, 2013, at 8:24 PM, peterpet wrote:
>
> [Dec 17 18:18:21] ERROR[12603]: chan_dahdi.c:10935 dahdi_ss7_error:
> Received message destined for point code 0x762 but we're 0x154a. D
=
> > coding=
> > framing=
> >
> > [2]
> > active=yes
> > alarms=UNCONFIGURED
> > description=Dynamic 'ethmf' span at 'eth3/00:50:C2:
> > name=DYN/ethmf/eth3/00:5
> > manufacturer=
> > devicetype=
> > location=
> > b
alaw=33-62
>
> On 12/05/2013 01:01 PM, Attila Domjan wrote:
> > In /etc/dahdi/system.conf did you define the sigchan like: mtp2 = 1?
> >
> >
> > On Thu, 2013-12-05 at 12:57 +0200, peterpet wrote:
> >> You catch me unprepared! I dont know! Explain me please w
In /etc/dahdi/system.conf did you define the sigchan like: mtp2 = 1?
On Thu, 2013-12-05 at 12:57 +0200, peterpet wrote:
> You catch me unprepared! I dont know! Explain me please what is mean
> kernel mode mtp2 ?!
>
> On 12/05/2013 12:48 PM, Attila Domjan wrote:
> > Are you
MngSLS: 1
> State: INSERVICE, UP
> STD Test: passed
> Got, sent :
> Inhibit:
> Changeover: NO
> Tx buffer: 0
> Tx queue: 0
> Retrans pos 0
> CO buffer: 0
> CB buffer: 0
> Last FSN: 7
> On 12/05/2013 12:24 PM, Attila Domjan wrote:
> > mtp3, isup timer values in configuration?
> >
> > On Thu, 2013-12-05 at 12:17 +0200, peterpet wrote:
> >> Configuration was checked 1000 times :)
> >>
> >> chan_dahdi:
> >>
> >>
WN!!!
> [Dec 5 12:14:45] WARNING[13364]: chan_dahdi.c:10094 ss7_linkset: MTP2
> link down (SLC 154888216)
> --- SS7 Down ---
> [Dec 5 12:14:47] ERROR[13364]: chan_dahdi.c:10935 dahdi_ss7_error: T7
> expired on link SLC: 1 ADJPC: 1606
> MTP2 link up (SLC 154888216)
> MTP2 link up
Hi, check the configuration, SLC -1214249800 is interesting
>
> [Dec 4 16:11:02] WARNING[7511]: chan_dahdi.c:10094 ss7_linkset: MTP2
> link down (SLC -1214249800)
> Changeover completed on link SLC: 0 PC: 1606 FSN: 6
> [Dec 4 16:11:02] ERROR[7511]: chan_dahdi.c:10935 dahdi_ss7_error:
> A
Some Siemens EWSD as STP doesn't send TRA...
AXE, AT&T send.
On Tue, 2013-08-13 at 13:56 +0200, Frederic Van Espen wrote:
> On 08/13/2013 01:20 PM, Attila Domjan wrote:
> > you didn't get TRA. Workaround:
> >
> > set mtp3_timer.t21 = 1 on linset 2.
>
Hi,
>-
>Adjacent SP PC: 12198 STATE: DOWN
>TRA: SENTT19: not running T21: not running
you didn't get TRA. Workaround:
set mtp3_timer.t21 = 1 on linset 2.
Regards,
Attila
>
>
> --
> __
In my system are used lot of custom cdr variables for many similar cases...
2013.04.21. 3:06, "Nyamul Hassan" ezt írta:
> Hi,
>
> We have an asterisk box with Digium 4xE1 card running libss7. All is
> well, except for a CDR issue about incoming calls whose DNID (local
> numbers) are not valid.
>
Hi!
just set the
mtp3_timer.t21 = 1
Regards,
Attila
On Tue, 2013-03-26 at 11:28 +0100, Pavel Troller wrote:
> Hi!
> I'm new in this list, I was "invited" by Kaloyan to post there in case
> of some questions or problems regarding the improved SS7 support in
> chan_dahdi.
> My first finding
gh
> Hi try without SIP user or IVR calls.
>
>
> On Thu, Feb 14, 2013 at 7:48 PM, Attila Domjan > wrote:
>
>> Just curious. Moved all of our SS7 interconnects to SIP.
>>
>> On Thu, 2013-02-14 at 16:15 +0200, Kaloyan Kovachev wrote:
>> > I am so g
Just curious. Moved all of our SS7 interconnects to SIP.
On Thu, 2013-02-14 at 16:15 +0200, Kaloyan Kovachev wrote:
> 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 abo
Its an very old and fixed bug.
We talked about it many times in this list.
On Thu, 2013-02-14 at 05:31 -0800, Marcus Vinicius wrote:
> Hello Jean,
>
> Thanks for your help.
> works fine with DTMF after the answer:
>
> exten => _10315,n,Dial(DAHDI/r3/${EXTEN},${RINGTIME},D(1))
>
> thanks a lot,
UP_EVENT_ACM:, or should the above solve the problem.
>
>
>
> Thanks,
> Dave George
> Teletone Inc.
> 561 674 3838
>
>
> -Original Message-
> From: asterisk-ss7-boun...@lists.digium.com
> [mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of A
I think it is the bug what I wrote it many times to this list, the
missing
p->proceeding = 1;
p->dialing = 0;
after the case ISUP_EVENT_ACM:, in static void *ss7_linkset(void *data)
A
On Wed, 2010-04-21 at 01:47 -0300, Rafael Prado Rocchi wrote:
> What asterisk version are you using?
> There'
On Fri, 2010-03-26 at 13:28 +0200, Kaloyan Kovachev wrote:
> Hi
> joining Matthew and thanks very much Attila! Without your patches and
> support there would be much less successful libss7 installations.
>
welcome! :)
> Are the hacks included in svn version? Can you point me to specific part
>
Hi!
I have some remark
- should add many members to asterisk channel structure to carry the SS7
attributes, eliminate the tons of SS7_* channel variables
- implement the new channel structure memebers to pass on other channel
types (SIP, IAX2)
- adding type flag on SIP/IAX2 channels to describe
t; callwaiting=yes
> usecallingpres=yes
> signalling=ss7
> group=5
> ss7type = itu
> ss7_called_nai=national
> ss7_calling_nai=national
> ss7_internationalprefix=00
> ss7_nationalprefix=0
> linkset = 5
> pointcode = 5176
> adjpointcode = 1203
> defaultdpc = 1203
> n
Hi,
you missed set up the timers
when you are using _X. like patterns, and you don't get end of pulse, on
isup_timer.digittimeout expiry will start the execution the dialplan.
set up all the timers, and avoid using _X. like patterns. When you know
the exact digitlengths use:
_0
and
Hi, you are right, thanks a lot. fixed in the svn.
The telco wanted to use and test but nothing happened...
On Fri, 2010-03-19 at 11:16 +0100, enrico mario wrote:
> Hi all,
> I think there's an error in libss7 (isup.h), exactly in the definition
> of ISUP parameter CUG Interlock Code. This paramet
On Fri, 2010-03-19 at 09:49 +0100, enrico mario wrote:
> Hi,
> setting "usecallingpres" to yes in chan_dahdi.conf did solve my
> problem. Now I can see the correct values for parameters in IAM
> message using my protocol analyzer: Address presentation restricted
> indicator: presentation restricte
d indicator: 0
> [ 40 14 ]
>
> -- Hungup 'DAHDI/3-1'
> Len = 16 [ a6 ae 0d 05 7c 53 4c 34 03 00 0c 02 00 02 81 a2 ]
> FSN: 46 FIB 1
> BSN: 38 BIB 1
> >[0] MSU
> [ a6 ae 0d ]
> Network Indicator: 0 Priority: 0 User Par
On Mar 17, 2010, at 1:07 PM, Attila Domjan wrote:
>
> > oops
> >
> > https://observer.router.hu/repos_pub/chan_dahdi/trunk
> >
> > On Wed, 2010-03-17 at 11:03 +0100, Attila Domjan wrote:
> >> I published it many times in the list with configuration examp
oops
https://observer.router.hu/repos_pub/chan_dahdi/trunk
On Wed, 2010-03-17 at 11:03 +0100, Attila Domjan wrote:
> I published it many times in the list with configuration examples:
>
> svn co https://observer.router.hu/repos_pub/libss7/trunk libss7
> svn co https://observer.rou
t;
> On Mar 17, 2010, at 12:46 PM, Attila Domjan wrote:
>
> > It is working in my version for me...
> >
> > On Wed, 2010-03-17 at 10:47 +0300, Pavel Piankov wrote:
> >> Hi,
> >>
> >> this did not help ether.
> >>
> >> libss7
It is working in my version for me...
On Wed, 2010-03-17 at 10:47 +0300, Pavel Piankov wrote:
> Hi,
>
> this did not help ether.
>
> libss7 1.0.2
> * 1.6.0.26
> dahdi 2.2.1
>
> it seems that ACM is sent in any case if a number has matching pattern in
> dialplan.
> is there any way to instruct
Hi, check the existence of the
p->dialing = 0;
in chan_dahdi.c, static void *ss7_linkset(void *data) after the
case ISUP_EVENT_CON:
case ISUP_EVENT_ANM:
and
case ISUP_EVENT_CPG:
near p->progress = 1;
On Thu, 2010-02-04 at 19:02 -0500, Dave George wrote:
> To have audio after the call is answ
Hi,
the ${HANGUPCAUSE} works fine in my box, however I don't use h
extension
Attila
On Wed, 2010-01-06 at 15:17 +0100, Krzysztof Drewicz wrote:
> Hi,
>
> how to obtain as much info from libss7 as possible when getting REL from
> telco:
>
> in debug i've got:
> Message Typ
try now please!
On Tue, 2009-11-24 at 16:09 +0100, Erik Wartusch wrote:
> Hi,
>
> Here is the answer I got from 02 czech telecom regarding the test I
> asked them using your latest code
> (https://observer.router.hu/repos_pub/libss7/sio_sios).
> Not successful from their side of view:
>
> "In "w
really willing to test and help but I don't want to make the telco
> angry as I have now several unsuccessful tests with them, dont know how
> long they will do that...
> Erik
>
> Am Dienstag, den 24.11.2009, 14:02 +0100 schrieb Attila Domjan:
> > please test this:
> &g
please test this:
https://observer.router.hu/repos_pub/libss7/sio_sios
On Tue, 2009-11-24 at 13:29 +0200, Kaloyan Kovachev wrote:
> On Tue, 24 Nov 2009 12:16:50 +0100, Attila Domjan wrote
> > Which q781 test case is failed?
> >
>
> in the attachment? it was 1.2 the one f
Please try this:
https://observer.router.hu/repos_pub/libss7/mtp2_t2_timer
On Tue, 2009-11-24 at 13:29 +0200, Kaloyan Kovachev wrote:
> On Tue, 24 Nov 2009 12:16:50 +0100, Attila Domjan wrote
> > Which q781 test case is failed?
> >
>
> in the attachment? it was 1.2
On Tue, 2009-11-24 at 13:29 +0200, Kaloyan Kovachev wrote:
> On Tue, 24 Nov 2009 12:16:50 +0100, Attila Domjan wrote
> > Which q781 test case is failed?
> >
>
> in the attachment? it was 1.2 the one for T2 timer
>
OK
> in mtp2_transmit() the flag is cleared aft
art over.
> > >
> > > If we start from different point in time it is 'send SIOS, send SIO start
> > > T2'
> > > this is exactly how it will behave when state is MTP_IDLE an mtp_setstate
> > > is
> > > called, but not if the s
DLE an mtp_setstate is
> called, but not if the state is MTP_ALARM which i could not find in the specs
> at all and i think it should be replaced with MTP_IDLE
>
> > On Mon, 2009-11-23 at 18:46 +0200, Kaloyan Kovachev wrote:
> > > On Mon, 23 Nov 2009 16:52:25 +0100, Attila Do
e it is 'send SIOS, send SIO start T2'
> this is exactly how it will behave when state is MTP_IDLE an mtp_setstate is
> called, but not if the state is MTP_ALARM which i could not find in the specs
> at all and i think it should be replaced with MTP_IDLE
>
> > On Mon, 2009
gt; > Good to hear about the progress.
> > > > > Would be really great If you can solve this MTP2 problem and merge to
> > > > > the trunk.
> > > > > My boss asked us allready to switch the two lines from libss7 to the
> > > > > commercial s
nd merge to
> > > > > the trunk.
> > > > > My boss asked us allready to switch the two lines from libss7 to the
> > > > > commercial ss7box (Sangoma) we also use for other carriers. Would be
> > > > > sad
> > > > > (an
> > > > Hi,
> > > > >
> > > > > Good to hear about the progress.
> > > > > Would be really great If you can solve this MTP2 problem and merge to
> > > > > the trunk.
> > > > > My boss asked us allready to switch t
gt; > > > Good to hear about the progress.
> > > > > Would be really great If you can solve this MTP2 problem and merge to
> > > > > the trunk.
> > > > > My boss asked us allready to switch the two lines from libss7 to the
> > > >
in t2_expiry.
I think it will solve the problem...
A
On Fri, 2009-11-20 at 17:15 +0200, Kaloyan Kovachev wrote:
> On Fri, 20 Nov 2009 16:01:07 +0100, Attila Domjan wrote
> > welcome, the mtp2 problem is not clear, I have not touch too much to
> > mtp2 layer...
> > what is
ease explain again.
On Fri, 2009-11-20 at 17:15 +0200, Kaloyan Kovachev wrote:
> On Fri, 20 Nov 2009 16:01:07 +0100, Attila Domjan wrote
> > welcome, the mtp2 problem is not clear, I have not touch too much to
> > mtp2 layer...
> > what is the current state of the mtp2???
> &
On Fri, 20 Nov 2009 15:36:49 +0100, Attila Domjan wrote
> > Hi, I made a test branch...
> >
> > svn co
> > https://observer.router.hu/repos_pub/libss7/accept_any_sls_tfp_tfa
> > libss7
> >
>
> Thank you very much again. You were faster than me, but while l
wrote:
> On Thu, 19 Nov 2009 12:32:58 +0100, Attila Domjan wrote
> > On Thu, 2009-11-19 at 13:10 +0200, Kaloyan Kovachev wrote:
> > Hi,
> > >
> > > Domian, do you know which ITU specs I should look at to confirm (or not),
> > > that
> > &
9.343] Message type: TRA
> [2009-11-13 15:44:49.343] [ 17 ]
> [2009-11-13 15:44:49.343]
>
>
> > On Wed, 2009-11-18 at 18:30 +0200, Kaloyan Kovachev wrote:
> > > Hello Domian,
> > > i have a similar (almost sure _the same_) problem:
> > >
o
> > > > the link remained as:
> > > > "Adjecent SP PC: XXX STATE: DOWN"
> > > > but
> > > > "State: INSERVICE, UP"
> > > >
> > > > unfortunately i can't find the full debug log right now, but w
was running)
>
> So sadly we got today again a negative result (test not passed) of the
> czech telco so we can not go live with the SS7 link...
>
> Kind Regards,
> Erik
>
>
> Am Mittwoch, den 11.11.2009, 16:10 +0100 schrieb Attila Domjan:
> >
Hi,
the mtp2 timer values wired in the code, you hacve to recompile.
A
On Mon, 2009-11-16 at 18:23 -0500, Paul Timmins wrote:
> Is there an easy way to adjust these without editing mtp2.h and recompiling?
>
> I seem to think there was a way to do this in chan_dahdi.conf but can't
> find anythin
MTP3 layer, which isn't
> implemented. But it is still required for libss7 to work as described
> in ITU.
>
> BR,
> K.
>
> В Срд, 11/11/2009 в 16:10 +0100, Attila Domjan пишет:
> > Hmm, it is interesting...
> >
> > static void t2_expiry(void * data)
> > {
&g
nning
> out.
>
> Any experience with that?
>
> Kind Regards,
> Erik
>
> Am Dienstag, den 10.11.2009, 11:30 +0100 schrieb Attila Domjan:
> > Hi, don't mix the libss7/asterisk(chan_dahdi) versions, there are some
> > api changes.
> >
> > My version (
Hi, don't mix the libss7/asterisk(chan_dahdi) versions, there are some
api changes.
My version (many additional features) are working from my svn.
Regards,
Attila
On Mon, 2009-11-09 at 17:35 +0100, Erik Wartusch wrote:
> Hello ss7 list!
>
> Installing libss7 and asterisk from
> http://svn.digiu
Hi sorry I missunderstand you. It must implemented via dialplan.
the hangupcause=256 means have to try dial again.
from my dialplan (ael):
CUR_TRY=0;
loop:
Dial(DAHDI/G2/${DIALNUM},90,L(${LIMIT_CALLTIME}${LIMIT_PLAY_WARNING})g);
NoOP(DIALSTATUS=${DIALSTATUS} HANGUPCAUSE=${HANGUPCAUSE});
CUR_TRY
Hi,
it is not implemented.
Regards,
Attila
On Mon, 2009-10-26 at 23:34 +0100, enrico mario wrote:
> Hi guys,
> I'm trying to test the automatic repeat attempt of libss7 (Attila's
> version), but I cannot get it to work. The message sequence I obtain
> is correct until the cic release (as described
Hi,
1st: you are right, clearning blocking for GRS is fixed in
chan_dahdi/trunk r59.
the segfault: I couldn't reproduce, could you send me the gdb output of
the core file?
Regards,
Attila
On Tue, 2009-10-20 at 20:49 +0200, enrico mario wrote:
> Hello guys,
> I've got a problem with my libss7 in
Hi,
which libss7 version are you looked into, I'm using mine and have tons
of incomming calls with SAM.
regards,
Attila
On Tue, 2009-10-20 at 20:32 +0800, Tian wrote:
> Hi guys,
>
> I am reading libss7 source code and I think I've found a bug in it,
> following is my description of the bug and m
7 version ?. I
> have problems with isup_timer.t8. I dont know who is the problem. Not
> working the TIMER 8.
>
> Thanks..
>
> - Original Message -----
> From: "Attila Domjan"
> To:
> Sent: Wednesday, September 30, 2009 11:24 AM
&g
think would be the
> equivalent for for
> sccp or tcap?
I don't know anough sccp and tcap waht implementation would be fine.
> what variables, just an example, of these layers would be in that
> struct ?
> regards,
>
>
> On Wed, Sep 30, 2009 at 9:03 AM, Attila Domjan
>
the most of are trivial or easy to find out from the code.
which members are not for you?
On Wed, 2009-09-30 at 08:29 +, Rony Ron wrote:
> Hi thanks for your reply,
> please can i have a per line comment ?
> regards,
>
> On Wed, Sep 30, 2009 at 7:26 AM, Attila
isup_call allocated for any isup session call and in my version for
circuit reset/(un)blocking sessions etc...
On Tue, 2009-09-29 at 23:47 +, Rony Ron wrote:
> Hi all,
> i am trying to develop sccp for libss7,
> i have began be trying to understand deeply the current code
> of isup; i have
t coming in Outbound Isup Call (Rajesh Mahajan)
> > 4. Re: Voice is not coming in Outbound Isup Call (Wasim Baig)
> > 5. Re: handling * and # of dialed number on the extension.conf
> > (Kaloyan Kovachev)
> > 6. Re: Voice is not coming in Outbound Isup Call (At
I assume ouccered by the missing p->dialing = 0; in chan_dahdi near
p->proceeding = 1; in case ISUP_EVENT_ACM: and case ISUP_EVENT_CPG:.
I wrote about it in many times in this list.
On Fri, 2009-09-18 at 11:41 +0530, Rajesh Mahajan wrote:
> Hi All.
>
> We are using Sangoma A104u Quad Card for SS
Hi,
try move the sigchan line after the networkindicator line.
A
On Tue, 2009-09-15 at 13:28 +0530, Rajesh Mahajan wrote:
> Dear All
>
> We are using following Software/Hardware.
> 1.asterisk-1.6.1.4
> 2.dahdi-linux-2.2.0.2
> 3.dahdi-tools-2.2.0
> 4.libss7-1.0.2
> 5.wanpipe-3.5.6
>
> Hardware:
>
the 3.4.1 wanpipe never crashed (yed), but I would like to use dahdi-2.2
with half-full buffer policy...
On Sat, 2009-08-08 at 12:50 -0500, resea...@businesstz.com wrote:
> I have done that already!! I am also optimistic that the new firmware i
> have installed works as till now i haven't managed
the v37 is for the A108 cards, for A102 the latest is the v34 and also
contains this fix. the firmwares are in the wanpipe and/or
ftp://ftp.sangoma.com/firmware
Last night I made another probe (A108D V37 firmware, dahdi-2.2.0.2,
wanpipe-3.4.4), the kernel panic came after the first ANM, so reverted
Hi,
the problem is:
hardhdlc=16
replace to
mtp2=16
hint: don't try wanpipe 3.4.4 this cause kernel panic with mtp2
wanpipe-3.4.1 + dahdi 2.1.0.4 works fine here
Regards,
Attila
On Mon, 2009-08-03 at 00:38 -0500, resea...@businesstz.com wrote:
> Hi Alejendro
>
> Have you managed to find out the
hi,
On Wed, 2009-07-01 at 10:32 -0300, Ing. Juan G. Dominguez wrote:
> Hello Attila, How are you doing ? Thanks for your answer. The problem of
> timers is finish.
> You know some distribution or SVN that support SAM in outcall ??
>
no because of asterisk limitation, in the upcomming (1.6.2 or
On Fri, 2009-06-05 at 14:22 +0200, marek cervenka wrote:
> > Yes it is needed my libss7 version.
> > This chan_dahdi.c tested and compile with asterisk-1.6.0.9.
> >
> > svn co https://observer.router.hu/repos_pub/libss7/trunk libss7
> > svn co https://observer.router.hu/repos_pub/chan_dahdi/trunk c
; sure where it was missing from.
>
> Anyway, thanks again for the help!
>
> Cheers,
> Zoltan
>
> Attila Domjan wrote:
> > I think it is the bug in the chan_dahdi, which is introduced by the
> > p->dialing not implemented proberly in the ss7 part of the chan_dah
I think it is the bug in the chan_dahdi, which is introduced by the
p->dialing not implemented proberly in the ss7 part of the chan_dahdi.
Check wheter exists p->dialing = 0; after the p->progress = 1;
in static void *ss7_linkset(void *data) function at the
case CPG_EVENT_INBANDINFO:
case ISUP_EV
On Tue, 2009-05-19 at 15:14 +0200, chamo wrote:
> hello,
> i have some problems with ss7 interconnect , with EWSD
>
> first, when i do outbound call to concrete channel , call is made, and
> it works no ok
>
> exten => _9.,1,Dial(DAHDI/6/${EXTEN:1})
> exten => _9.,n,Hangup
>
> but if i dial with
Hi,
added digit timeout if the incomming call doesn't have en of pulse
digit.
I thinking about using ss_thread, but sorry it is a nightmare...
So I started a timer in the libss7.
to setup the timer:
isup_timer.digittimeout = 5000
(5s)
btw I suggest avoid using . -ending pattern in the dialplan
Hi,
Fixed the sls in mtp3 messages. Have to restart counting links per STP.
(Q.704 2.2.4, last paragraph).
Thanks for Bruno Rodrigues helping the testing.
Attila
signature.asc
Description: This is a digitally signed message part
___
--Bandwidth and C
I merged an tested many patch from 1.6.0 branch.
Matthew: intoduced an ss7 bug in chan_dahdi by r183327
1.6.0 svn (and 1.6.0.9).
some inband indication won't be passed to SIP and others.
Missing the
p->dialing = 0;
after the p->progress = 1;
when we get ACM/CPG
(it is fixed in my svn)
Regard
On Mon, 2009-04-06 at 00:04 +0200, Bartosz Supczinski wrote:
> > you have not set up the timers...
> > the suggested itu values are:
>
> does not work...
>
yes, but its not the fault of the libss7/chan_dahdi
You did not get SLTA for SLTM.
you have to see something like this:
Len = 20 [ 85 86 11
Added now the T21 mtp3 timer.
If we decided to restart mtp3 and the other party has t19 still running
(won't send any TRA), we will accept the traffic, if T21 expired.
suggested itu value:
mtp3_timer.t21 = 63000
Regards,
Attila
On Fri, 2009-04-03 at 04:46 +0200, Bartosz Supczinski wrote:
> > 1th
Please send the full starting dump
(SLTA/SLTM, TRA )
Regards,
Attila
On Fri, 2009-04-03 at 04:46 +0200, Bartosz Supczinski wrote:
> > 1th: try my libss7/chan dahdi branch
> > 2nd: define and use prefixes to set nai
>
> not work correctly ...
>
>
> Len = 14 [ 81 83 0b 85 43 9c 00 17 11 00 1
Added ISUP T8 and T24 continuity related timers.
The suggested ITU values are:
isup_timer.t8 = 1
isup_timer.t27 = 24
We don't loop the previous circuit if it is used (race condition, we
allocate the cic and the other party ask a loop on it in IAM on the next
circuit).
the new continuity
Hi,
chan_dahdi now send rel(17) for Busy and rel(42) for Congestion instead
of playing inband indication
if
priindication = outofband
is set.
svn co https://observer.router.hu/repos_pub/chan_dahdi/trunk chan_dahdi
svn co https://observer.router.hu/repos_pub/libss7/trunk libss7
Regards,
Attila
nai is set to
> national in the config or they are used for the outbound calls only?
>
> On Wed, 11 Mar 2009 12:18:35 +0100, Attila Domjan wrote
> > Use another char (like M, P) for prefix, if your provider so idiot using
> > prefix in ss7
> >
> > On Wed, 2009-03-11 at
Use another char (like M, P) for prefix, if your provider so idiot using
prefix in ss7
On Wed, 2009-03-11 at 12:55 +0200, Kaloyan Kovachev wrote:
> Hello,
> I have repeated the results with two Asterisk via dynamic:
>
> the first Asterisk is configured as nai dynamic for both called and calling,
the new urls:
svn co http://observer.router.hu/repos_pub/chan_dahdi/trunk chan_dahdi
svn co http://observer.router.hu/repos_pub/libss7/trunk libss7
Regards,
Attila
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___
--Bandwidth and
Hi Matthew,
Just wanted let you know the UPC-Telecom also accepted our ss7 stuff
(They have Ericsson AXE switches)
Regards,
Attila
> You can watch the commit log for the svn branch that I'm using for it :-)
>
> I'm just trying to get to a point where I'm ready to have people start
> testing wi
than in:
static FUNC_SEND(called_party_num_transmit)
parm[1] = 0x1 << 4;
replace to
parm[1] = (0x1 << 4) | (0x1 << 7);
On Wed, 2009-01-21 at 09:57 +0100, Krzysztof Drewicz wrote:
> 2009/1/21 Attila Domjan
> >
> > hi,
> >
> > in isup.c
> >
hi,
in isup.c
in
static FUNC_SEND(calling_party_num_transmit)
before the return:
parm[1] |= 0x80;
Regards
Attila
On Wed, 2009-01-21 at 09:01 +0100, Krzysztof Drewicz wrote:
> Hello,
> for some tests i need to setup a call with NI=1
> --OPTIONAL PARMS--
> Calling Party Number
Don't forget the correct handling of the circuit group messages, may
have to split the GRS, CGB, etc. and concat the answers for them (GRA,
CGBA...)
When I have time I'll test your new stuff.
Regards,
Attila
>
> Well, to have a fully redundant setup, you would have separate boxes
> terminating
+ many fix, to pass the itu-t isup tests.
patch uploaded, but needed some small work yet...
http://bugs.digium.com/view.php?id=13495
Regards,
Attila
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--Bandwidth and Colocation P
Yes it will
I'm working many improvements/fixing on libss7, I hope today will be
ready for testing.
my cli is:
*CLI> ss7 block cic
Usage: ss7 block cic
Sends a remote blocking request for the given CIC on the
specified linkset
Regards,
Attila
On Tue, 2008-09-16 at 11:20 +0200, Martin
:
> On 09/02/08, Attila Domjan wrote:
> > Maybe...
> >
> > linkset1 connecting to SW1
> > linkset2 connecting to SW2
> >
> > the signalling path to SW1 is via STP1 and STP2
> > the signalling path to SW2 is via STP1 and STP2
> >
> > 1st E
der the same linkset. As long
> as your PC is the same this will work. Also looking at your conf it
> seems that you have not setup your CICs correctly.
>
> --
> Markus
>
>
> On Sep 2, 2008, at 11:35 AM, Attila Domjan wrote:
>
> > The telco wants this configura
> channel.
>
> --
> Markus
>
>
> On Sep 1, 2008, at 2:00 PM, Attila Domjan wrote:
>
> > Hi,
> > Is possible to set up the following example with libss7?
> >
> >
> >
But, have to implement the network routing number nature of address
(=8), like ss7_internationalprefix it could be ss7_networkroutingprefix.
On Tue, 2008-09-02 at 07:38 +, Rony Ron wrote:
> Hi,
> imho you can do it with call forward,
> you receive the number
> you check the database if the num
Hi,
asterisk can't be an STP.
On Tue, 2008-09-02 at 03:57 -0300, Virmones Pereira wrote:
> Hi,
>
> I would like to use asterisk with SS7 as a STP for Number Portability
> GW, the idea of the system is follow:
>
> When the SS7 central(Ericsson AXE) receive the call this should be
> route to the A
Hi,
Is possible to set up the following example with libss7?
TELCO
STP1(pc:2)- SW1(pc:4)
/ | \ /
/| \ /
my aserisk(pc:1) |\
\| / \
\ |
Hi Matthew,
I looked into docs/sorcecode and I have some questions about libss7.
Is it support (ITU):
- signalling link inhibition/uninhibition Q.782 7.1.1, 7.1.2, 7.2.1,
7.2.2, 7.6.1
- dual seizure handling Q.784 2.1.1, 2.1.2
Regards,
Attila Domjan
On Tue, 2008-08-12 at 12:37 -0500, Matthew
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