Florian,
it looks like you follow this list not for a long time. chan_ss7 devs
are also subscribed to it. While this question is asterisk ss7
specific, there is a reason to ask here. If someone experienced the
given, he may respond.
2011/9/2 :
> Wasim and general audience,
>
>
>
> I felt addresse
Look for a patch by Torrey Searle, in the recent mailing list posts, it should
apply to latest Atilla SVN
On Thursday 16 September 2010 17:22:18 Pavel Piankov wrote:
> Hello everybody,
>
> anybody knows if Attila's patches exist for asterisk 1.6.2.5?
>
> thanks
> Pavel
--
it working
>
> If Atilla sees this I give him permission to add it to his SVN :-)
>
> Torrey
>
> On 9 September 2010 19:10, Anton VG. wrote:
>> Are you sure you actually compiled Domjan's chan_dahdi? I'm getting errors
>> while trying compiling for 1.6
wrote:
> Did you run into compilation issues? I just compiled Atilla's
> chan_dahdi.c yesterday with 1.6.2.11 without any issue...
>
> Torrey
>
> On 8 September 2010 23:06, Anton VG. wrote:
> > Have Anyone used (updated) Domjan libss7+latest asterisk 1.6.2.x?
&
ithout any issue...
>
> Torrey
>
> On 8 September 2010 23:06, Anton VG. wrote:
> > Have Anyone used (updated) Domjan libss7+latest asterisk 1.6.2.x?
> >
> > Just looked to latest Domjan svn - the chan_dahdi.c
Have Anyone used (updated) Domjan libss7+latest asterisk 1.6.2.x?
Just looked to latest Domjan svn - the chan_dahdi.c is still for 1.6.0.x.
Any plans to update?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital
Thanks Matthias!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7
schannel =>
> firstcic => 97
> enabled => yes
>
> [host-SJO_P0BX_01]
> enabled => yes
> opc => 0x601
> dpc => ls1:0x103
> links => l1:1,l2:2,l3:3,l4:4
> globaltitle => 0x01, 0x01, 0x00, 4702636074
> ssn => 0
>
>
Also, at first, it is advised to read this list archive,
before asking the same question again and again. Domjan
several times posted a link to his SVN, which is stable and
used by many of us.
On Friday 31 July 2009, Saeed Ahmed wrote:
> Hi Tian,
>
> As advised by developers libss7 latest trunk
I experience this issue since 0.8x ... switched to libss7
because of it.
On Tuesday 24 February 2009, Mathias Doerr wrote:
> Hello,
> I`m using chan_ss7-1.0.10 with Digium TE410 card . Very
> often is one way audio where the called party can`t hear
> the callee, I `ve seen this problem was also
On Wednesday 18 February 2009 15:10, Kaloyan Kovachev wrote:
> On Wed, 18 Feb 2009 09:40:01 +0100, Attila Domjan wrote
>
> Isn't it a good idea to make a new branch for the IP part
> and leave the changes from bug 13495 alone?
> I do use a specific revision of the branch, because i
> want to have t
On Thursday 04 December 2008 03:47, Matthew Fredrickson
wrote:
> http://svn.digium.com/svn/libss7/team/mattf/bug13495
>
> http://svn.digium.com/svn/asterisk/team/mattf/bug13495
Matthew, with the URL you provided, there is no chan_dahdi
patches, everything downloaded is libss7, am i doing
somet
Guys, just confused which version should I try to be at the
most complete one, Should I still check with Domjan SVN or
all of the patched have been ported to the
http://svn.digium.com/svn/libss7/team/mattf/bug13495
http://svn.digium.com/svn/asterisk/team/mattf/bug13495
?
I'd like to notice that I have LIBSS7+Domjan patches working
for me in production (light load) for a month now. Will try
updated versions soon. Thank you Domjan and Matt for such a
great stuff.
On Thursday 04 December 2008 03:47, Matthew Fredrickson
wrote:
> Hey all,
>
> Thanks to a Adomjan on
waw, svn access is really great, thanks Domjan!
On Saturday 22 November 2008 00:45, Domjan Attila wrote:
> so they are there:
> svn co http://87.242.0.27/repos/trunk/chan_dahdi/
> svn co http://87.242.0.27/repos/trunk/libss7/
>
> todo:
> - make ss7linktest work again
> - documentation of the new f
e dialplan to send ACM.
;ss7_explictacm=yes
On Saturday 22 November 2008 12:36, Anton wrote:
> Look for expliciatanm setting in chan_dahdi.conf
>
> On Friday 21 November 2008 23:20, Rafael Visser wrote:
> > Hi guys
> >
> > I have an asterisk 1.6 working as an ivr co
Look for expliciatanm setting in chan_dahdi.conf
On Friday 21 November 2008 23:20, Rafael Visser wrote:
> Hi guys
>
> I have an asterisk 1.6 working as an ivr conected to an
> ericsson switch with libss7 1.0.2, and it's application
> only querys a data base and plays some messages.
>
> For every c
Hey Domjan,
Your patches great, I'm already using them on one of my live
servers. I've noticed one issue with them in the tests, but
it's not too critical.
Regards,
Anton.
On Tuesday 18 November 2008 01:32, Domjan Attila wrote:
> Hi,
> I hope tomorrow will be fixed my
I can go to IRC when needed. Just tell me when.
Regards,
Anton.
On Monday 10 November 2008 21:09, Matthew Fredrickson wrote:
> Anton wrote:
> > Hi Matthew,
> >
> > One of the ss7 bugs I've discovered on quick tests, and
> > it's critical to normal function
is no
activity in BT since 24th of October?
Regards,
Anton.
On Sunday 09 November 2008 20:32, Anton wrote:
> Just confirming the same bug behavior while testing
> libss7 to libss7 , and libss7 to telco switch.
>
>
>
> ___
> --Bandwid
Just confirming the same bug behavior while testing libss7
to libss7 , and libss7 to telco switch.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lis
Hi Matthew,
One of the ss7 bugs I've discovered on quick tests, and it's critical to normal
functionality:
short description: ss7 does not release channels on the other side channel
reset/failure.
It means, when there are some amount of calls going through the system, and
remote switch just
Hi Anders,
Glad that you answers, I was not expecting that :)
My response is below, inline.
On Sunday 09 November 2008 14:35, Anders Baekgaard wrote:
> Hello Anton,
>
> You write that chan_ss7 is intentionally filled with
> bugs, and you mention one bug, about the one way audio
>
DICEA "supports" chan_ss7 in a very bad way. There is
a "FIXED" version of it, and free version INTENTIONALLY
filled and left with bugs in different places. One of
the "intentional" bugs - accidental one way audio after a
while if usage on certain CIC's , another - improper
attaching to the /d
As for DAHDI support, Sangoma should have it's support in
the latest beta drivers. I did not try myself though, just
got this info recently.
On Monday 29 September 2008 06:59, mark morreny wrote:
> Hi,
>
> I have two Sangoma 101 cards that I would like to use to
> hook up two Asterisk. I tried w
k through the libss7 code I could
> only see were it addressed and iam with cot test type 1
> not cot test type 2.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
> Of Anton Sent: Thursday, September 25, 2008 12:16 AM
> To: asteri
I have to note that a while ago one edge of the problem was
in IAX connections between systems - so IAX was giving
oneway for a certin amount of calls - and switching to SIP
resoved it - but now it might be just chan_ss7 related.
On Wednesday 17 September 2008 21:43, Antoine Megalla wrote:
> He
Time from time I'm having one-way audio on the SS7 lines -
even rarely than 20% - maybe 1% of calls - maybe it's the
same issue. If anyone also noticed the same problem or
found a solution or workarround would appreciate for
letting know.
On Wednesday 17 September 2008 21:43, Antoine Megalla w
Can you share your modifications?
On Saturday 13 September 2008 00:47, Domjan Attila wrote:
> Hi,
> I implemented many circuit (circiut group still progress)
> isup timers and modification for passing itu q.784 test
> (like don't sent rel when got iam on unconfigured cic and
> the many other abnor
Use OSLEC - others just sux in comparision to OSLEC.
On Tuesday 09 September 2008 15:08, olivier taylor wrote:
> Ok,
>
> another question.
> They are many possibilities for echo canceller (jpah,
> kb1, mg2, sec2, and sec)
>
> Wich is the best, more efficient?
> Is it also possible to use Oslec
Segfaulted for me yesterday, on trying 1.0.94, so returned
back to 1.0.10
On Sunday 07 September 2008 21:38, Grant Arix wrote:
> Hi all,
>
> I wanna know if somebody has ever tried chan_ss7 beta ?
>
> king regards.
>
>
>
>
> ___
> --Bandwidth and Coloca
Hello!
Just interested, if anyone have/had a similar problem with
Sangoma A108 card. When I use ALL 8 ports on it - I loose
signalling on all plugged ports (chan_ss7) - if I use up to
7 ports - it's OK. Use of TDM ports - regardles of
signalling - it could be PRI or SS7 - I do plug 8th port
a
Ususally you've got syntax misconfiguration in
extensions.conf - see _X. or X. - behave differently.
On Wednesday 03 September 2008 13:51, Nguyen Trung Thanh
wrote:
> Dear all,
>
>
> I am setting SS7 link using on chan_ss7. I have one link
> for signal, one link for voice. I also could make a
>
Hey Matthew, good to hear that!
If you are willing to write the driver compatible for HUAWEI
MGCP/MEGACO (telco grade) gateways, I have 1 port gateway
here, and can donate it for this task.
On Thursday 14 August 2008 20:25, Matthew Fredrickson wrote:
> Anton wrote:
> > As SS7 MGCP you
As SS7 MGCP you probably mean MEGACO protocol?
Unlikely this could be seen in asterisk anytime soon...
On Thursday 14 August 2008 11:11, voip me wrote:
> Dear Matthew,
>
> What do you think , how could one improve it then, you
> think chan_dahdi should speak MGCP or any request show
> come up to
There is a number of precompiled packages, try different
types
On Thursday 07 August 2008 16:37, olivier taylor wrote:
> hello,
>
> I have tried many of them.
> My architecture is I686 and all the binaries I've tried
> gave me a core dump...
>
> br,
>
> Olivier
>
>
> Rony Ron a écrit :
> Hi
Look for an IPP version than.
On Thursday 07 August 2008 11:21, olivier taylor wrote:
> Hello all,
>
> I have two servers installed with the latest SS7
> rellease (asterisk 1.6.0 and chan_dahdi).
>
> It works perfectly, thanks to the list.
> But now, I need to install G729 on these servers but
Hm, good, not a long time ago there was no any info on the
dice site on anything above 1.0.0 ж)
On Wednesday 06 August 2008 13:49, Tusar wrote:
> http://www.dicea.dk/download/NEWS-1.0.93-beta.txt
>
> On Wed, Aug 6, 2008 at 2:22 PM, Anton
<[EMAIL PROTECTED]> wrote:
> > Anyon
Anyone aware what would be the changelog for a new beta?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7
pri1.4.3
> I am using a Digium TE120P.
>
> Thanks.
>
> --- On Wed, 6/8/08, Anton <[EMAIL PROTECTED]> wrote:
> > From: Anton <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk-ss7] chan_ss7 - unstable link
> > To: asterisk-ss7@lists.digium.com, [EMAI
Versions of asterisk zaptel and chan_ss7?
On Wednesday 06 August 2008 11:27, Low Yu Siang wrote:
> Hi all,
>
> After running for days(~300 incoming calls), the link
> became unstable. Whenever a call is coming in, it throws
> the following error messages. This call is accepted
> properly but all o
is set at 30
> seconds, are you suggesting that I should modify the code
> so that it is >120 seconds?
>
> Thanks.
> Yu Siang
>
> --- On Mon, 21/7/08, Anton <[EMAIL PROTECTED]> wrote:
> > From: Anton <[EMAIL PROTECTED]>
> > Subject: Re: [asterisk
e also appearing. Currently I see in the
> chan_ss7 code that the timeout value is set at 30
> seconds, are you suggesting that I should modify the code
> so that it is >120 seconds?
>
> Thanks.
> Yu Siang
>
> --- On Mon, 21/7/08, Anton <[EMAIL PROTECTED]> wrote:
> >
I have similar issue with connecting to ZTE and HUAWEI
switches. GRA comes not imediatelly, but after 1-2 minutes.
Maybe you have the same issue. Did you wait long enough?
On Saturday 19 July 2008 19:39, Low Yu Siang wrote:
> Hi all,
>
> Thanks to the generous help from Mr. Pawel Ratajewski, I
>
On Wednesday 16 July 2008 17:43, Antoine Megalla wrote:
Hi, It Will be interesting setup.
I'm using SANGOMA A108 hardware and do run 22xE1 per a
single server A108x2+A104+A102, though, a little patching
of the chan_ss7 will be necessary - dig the mailing list, I
had posted the necessary change
NGOMA 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,
> >I can only comment on it workin
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
As I understand this would require some patching, since
there is no native functionality yet? Actually call
divert/forward is a very required feature... Possibly any
extra dialplan function required, like Forward() ?
On Thursday 10 July 2008 02:02, Alan McMillan wrote:
> I have successfully im
I do use 1.0.10 in production.
On Tuesday 01 July 2008 15:32, Suhaib Mehyar wrote:
>again plz if u can advice of a tested version of
> asterisk and zaptel with chan_ss7-1.0.10.
>
> many thanks.
>
>
> From: [EMAIL PROTECTED]: [EMAIL PROTECTED];
> [EMAIL PROTECTED]: Tue, 1 Jul 2008
> 10:28:
first try upgrading your chan_ss7 version. 0.9 is quite old.
On Monday 30 June 2008 13:22, Suhaib Mehyar wrote:
> dears,
>
>
> am using chan_ss7 and i was surprised by getting the
> error shown down, in someday everything was wroking fine
> till this message start to appear. kindly note that i
>
And also to create/delete/modify trunk, span. etc :)
Even for PRI, which is in asterisk for years the
functionality is not there :(
On Monday 05 May 2008 12:50, Christopher Bautista wrote:
> Hi Everyone
>
> I badly need a tool to restart an E1 independently
> without having to restart everything
I do use sangoma A108 card with chan_ss7 - if you have HW
echocancellers on board - they does bother. Trying
cancelling an echo on sigchannel. dig mailing list for
sangoma settings to deal with this.
On Thursday 24 April 2008 00:36, Barry O'Donovan wrote:
> Hi folks,
>
> I'm trying to use eithe
Mobile operators operates own echocancellers on every
channel, but not landlines. I operate chan_ss7 witch
SANGOMA cards with HW echocanceller - and it helps on
landlines.
On Friday 11 April 2008 06:55, Christopher Bautista wrote:
> Hi guys
>
> Update:
> > PSTN -> Asterisk -> SIP extensions
>
>
Better to the list, I'm iterested in it's behaviour either.
On Wednesday 02 April 2008 15:40, marek cervenka wrote:
> hi,
>
> there is new version with jitter buffer
> http://www.dicea.dk/download/chan_ss7-1.0.10.tar.gz
>
> please send feedback (private to cervajs at fpf.slu.cz or
> public to the
Cool, this is not noted on dicea site...
On Wednesday 20 February 2008 21:16, marek cervenka wrote:
> On Wed, 20 Feb 2008, Antoine Megalla wrote:
> > Hi,
> >
> > I am using chan_ss7 1.0.0 with success, where I pass
> > SIP calls to SS7 switch, however I am facing a nagging
> > problem that prevent
If you provide a backtrace - it might be possible to get the
point of failure. How to do backtrace with gdb and how to
compile your asterisk with debugging info you will have to
find yourself or maybe someone else on the list will brief
you on this.
On Sunday 13 January 2008, Tusar wrote:
> Hi
non-blocking: "
"%s.\n", strerror(errno));
goto fail;
}
return fd;
fail:
return -1;
}
On Sunday 09 December 2007, Anton wrote:
> Hello Anders!
>
> Still cannot use chan_ss7 with over 8 e1s since chan_ss7
> tries to access to /dev/zap/channel
/264: No such file or directory
Regards,
Anton.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit
Looks like 3.3x branch of wanpipe still needs explicit
disabling of echocancelling on the sigchannel so it does
not consist the TDMV_HWEC_PERSIST_DISABLE parameter
On Thursday 06 December 2007, Goke Aruna wrote:
> However, if you download the sangoma driver version
> 3.2.1.8, it will work with e
Hi Anders!
Could you please share with us some info, just
if you aware of any production use of chan_ss7 1.0.0 and
possibly amount of E1 in single installation and maximum
amount of E1 in the system?
Thanks in advance!
Anton.
On Tuesday 27 November 2007, Anders Baekgaard wrote:
> Dicea
In Pre 3.0 sangoma wanpipe drivers you need to add
TDMV_HWEC_PERSIST_DISABLE = YES to avoid echocanceler
working all of the time and so corrupting digital data
sangoma tells that in 3+ this is fixed but i did not try it.
On Monday 03 December 2007, Goke M Aruna wrote:
> On 12/3/07, Anton <
Try to switch off the hardware echo canceler
TDMV_HWEC = YES
On Monday 03 December 2007, Goke Aruna wrote:
> I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0
> on FC7 all went okay. using sangoma a104dx on both
> machine.
>
> I followed the write up on
> http://www.voip-info.org/wik
Dear Anders,
You can't imagine what the excellent news is this!
Little question btw, does the 1.0.0 support over 8 physical
E1 in single PC? (0.9 and prior does not)
My best wishes,
Anton.
On Tuesday 27 November 2007, Anders Baekgaard wrote:
> Dicea has released a major update of
Bruno's changes for asterisk 1.4 is good, but you need also
to add Denis Smirnov's changes for ZAP style addressing -
for me it works as a combined version. And RBT may be
usefull for Erricson switch only - since for other switches
(ZTE and HUAWEI for me) it works without this change fine.
On
Nico,
Do you think it's time to give libss7 another try? My last
test (3-4 month ago) gave terrible results - links did not
restart automatically , channel was dying accidently and
unexpectedly and so on.
Anton.
On Wednesday 14 November 2007, [EMAIL PROTECTED] wrote:
> I used chan
Hi Hoai-Anh Ngo-Vi!
What did you port? Any code changes? improvements?
On Tuesday 13 November 2007, Hoai-Anh Ngo-Vi wrote:
> Hi,
>
> I've ported chan_ss7 to use in production environment.
>
> With 8 E1s in Germany
>
> cheers
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[
BTW, I just had a one-way audio situation on one ss7 link
while using SIP.
On Friday 09 November 2007, Anton wrote:
> I could speculate that IAX in conjunction with chan_ss7 -
> leads to that behavior - breaks something or so. - Try
> SIP... And please let know if behavior reappea
t when any CIC in SS7 link gets this
> "strange" state, even looped calls SS7-SS7 through this
> CIC have one way audio - incoming, outgoing audio
> direction is silent ...
>
> - Dawid
>
> 2007/11/8, Anton <[EMAIL PROTECTED]>:
> > Do you use IAX on this se
Do you use IAX on this server? If so try SIP instead, let
know here if so...
But a some noticed this behavior before, including me, and
now I'm not sure what was the reason, IAX or chan_ss7
On Thursday 08 November 2007, Dawid Kerad wrote:
> Helo,
>
> I have a problem with one way audio using c
IMHO encouraging people to not formulate a question
properly, by doing THEIR job, is not a good approach.
shantha: properly asked question is already resolved on 50%.
Learn properly asking questions by giving enough info.
On Friday 28 September 2007 11:08, sri shantha wrote:
> i need to transfe
The best idea is to hire a professional masseur.
On Friday 28 September 2007 09:17, sri shantha wrote:
> DO u have any idea for how to transfr the massage by
> using linux
>
> Indra Sri Shantha Abeysinghe
> [EMAIL PROTECTED]
> [EMAIL PROTECTED]
> 0724-301904
>
>
>
>
>
>
> _
Kristian,
Any hope for next chan_ss7 version? Any hope for Native
Sifira 1.4 chan_ss7 support?
BTW: JB in 1.4 greatly reduces amount of write buffers full
errors.
Regards,
Anton
On Saturday 22 September 2007 11:18, Kristian Nielsen wrote:
> "Mr.Surender Reddy" <[EMAIL PR
receiving they have
> invalid BSNs and FIBs.
> Could anyone please help.
> Any clue will be appriciated.
>
> Regards,
> Arvind
--
Sincerely,
Anton V. Gnitko
Technical Director
ISP Eastera Co. Ltd.
(w) +992 372 270101
(w) +992 372 213627
(cell) +992
IC=62.
> [Mar 21 10:00:31] NOTICE[14057]: l4isup.c:768 t1_timeout:
> T1 timeout (waiting for RLC) CIC=60.
> [Mar 21 10:00:31] NOTICE[14057]: l4isup.c:768 t1_timeout:
> T1 timeout (waiting for RLC) CIC=54.
> [Mar 21 10:00:31] NOTICE[14057]: l4isup.c:768 t1_timeout:
> T1 timeout (wa
Though the same config works with another telco. But anyway
Some kind of channel allocation parameters than might be
usefull
On 20 March 2007 22:30, Anton wrote:
> Today I've found a very bad behaviour with chan_ss7 -
> while operating it on the 2xE1 link EVERYTHING on the 2nd
> E
ion on
> '[EMAIL PROTECTED]' of
> Request 102: Match Found
> Mar 20 19:02:30 DEBUG[5694] chan_sip.c: SIP response 200
> to standard invite
>
>
>
> BR
> Ercan
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECT
Today I've found a very bad behaviour with chan_ss7 -
while operating it on the 2xE1 link EVERYTHING on the 2nd E1
runs one-way audio. It means that while using even-mru
we're going to have every call after 15th to be oneway.
Someone have been reporting that >15th call have one-way
audio. 3x h
Just put Dial(SIP...) or Ringing than Dial there... Don't
put Answer in dialplan if you do not mean it.
On 19 March 2007 03:51, Mitul Limbani wrote:
> Hello Ercan,
>
> Quoting Ercan Yücebas <[EMAIL PROTECTED]>:
> > Dear All
> >
> > Is there other ways to not answer the channel in the
> > dialpla
0.8.4 has the same problem.
On 8 March 2007 15:27, Anton wrote:
> Just looked through the mailing list - Nico
> [EMAIL PROTECTED] reported that a while ago -
> Confirming - I met the same situation - chan_ss7 or
> asterisk (atleast up to 1.2.12.1) have the channel
> allocation
Your telco may not allow calls with no callerid - ask them
On 16 March 2007 11:52, umar tarar wrote:
> you can logically avoid this problem by giving something
> (e.g. a constant number/digit/alphabet) in callerid
>
> On 3/15/07, Pawel Ratajewski (Forweb) <[EMAIL PROTECTED]>
wrote:
> > hi,
> >
>
:19, Anton wrote:
> Does anyone experiencing a situation, when loaded
> chan_ss7 0.9 after a while of operation (for me 2 days) -
> stopped accepting calls - while having all of the
> channels free - I'm getting congesion - "no idle circuit
> found" - aster
Does anyone experiencing a situation, when loaded chan_ss7
0.9 after a while of operation (for me 2 days) - stopped
accepting calls - while having all of the channels free -
I'm getting congesion - "no idle circuit found" - asterisk
reload cures the situation.
_
Anders,
Any chance to know when are you planning to release a next
chan_ss7 version? As you may know chan_ss7 0.9 does not
work with asterisk>1.2.14 and zaptel>1.2.12 . Hopefully
that will not be an issue for a new version.
Regards
Anton.
On 23 February 2007 13:56, Kristian Nielsen
On a PC with a little memory amount (128m) - seems there is an issue in 0.9
(0.8.4 too) - i've been getting a stable segfault in the line
while processing GRA on some of the CIC.
0.9
struct ss7_chan* pvt = event->isup.link->linkset->cic_list[isup_msg.cic];
__
In the scheme A <--> B, description given below describes
that is CIC is blocked by side A, that side B cannot
initiate calls to A, but A can initiate to B unless B also
sent a blocking request.
(http://www.asknumbers.com/SS7ISUPMessages.aspx)
Quoting ---
Blocking (BLO) message sent only for ma
ription or
behavior.
Regards,
Anton.
On 23 February 2007 13:56, Kristian Nielsen wrote:
> Anton <[EMAIL PROTECTED]> writes:
> > In my case? when i do have 1way audio, it's always in
> > the IN direction. I mean in the scheme
> >
> > --SIP-- - the side
>
I had yesterday the case again.
In my case? when i do have 1way audio, it's always in the IN
direction. I mean in the scheme
--SIP-- - the side
cannot hear the - but users hears PSTN.
What's in your case? Any other behaviors?
On 23 February 2007 13:07, [EMAIL PROTECTED] wrote:
> I don't
ssage -
> >>> From: <[EMAIL PROTECTED]>
> >>> To:
> >>> Sent: Monday, February 19, 2007 11:03 AM
> >>> Subject: Re: [asterisk-ss7] Which Versions
> >>>
> >>> | Which Version of chan_ss7 do you use and which
> >>>
sion of chan_ss7 do you use and which
> >> | Distribution, Kernel and architecture is running on
> >> | your Server?
> >> |
> >> | Thanks for the info
> >> |
> >> | Nico
> >> |
> >> | On Sat, 17 Feb 2007, Mr.Surender Red
PROTECTED]>
wrote:
> >> I forwarded it now to sifira, i will give you an
> >> update if there are some news.
> >>
> >> Nico
> >>
> >> On Thu, 15 Feb 2007, Anton wrote:
> >> > Why don't you forward the bug report to someone from
Why don't you forward the bug report to someone from Sifira,
they should be interested.
On 15 February 2007 13:32, Christian Lueger wrote:
> Hello,
>
> I am having the same problem.
>
> Also with chan_ss7-0.9.
> My asterisk ist 1.2.7.1 and my Zaptel is 1.2.5.
> Still found no solution for this.
>
absolutely.
On 12 February 2007 14:26, sri shantha wrote:
> R u sure???
>
> Indra Sri Shantha Abeysinghe
> [EMAIL PROTECTED]
> [EMAIL PROTECTED]
> 0724-301904
>
>
> - Original Message
> From: Anton <[EMAIL PROTECTED]>
> To: asterisk-ss7@li
www.google.com
On 12 February 2007 14:16, sri shantha wrote:
> hai i need to know how to establish the E1 link
> regards
>
> Indra Sri Shantha Abeysinghe
> [EMAIL PROTECTED]
> [EMAIL PROTECTED]
> 0724-301904
>
>
>
>
>
>
> _
>_
Guys,
There is OpenSS7 project doing progress, I see many releases
coming this year and it's written that most parts are in
production release. Does anyone have any
experience/impression with OpenSS7 that could be shared?
___
--Bandwidth and Colocati
On 30 January 2007 19:16, Kristian Nielsen wrote:
> "Jorge Churio" <[EMAIL PROTECTED]> writes:
> > Current asterisk SS7 implementations (both chan_ss7 and
> > libSS7) only supports fully associated signaling that
> > means signaling links must be in the same E1/T1 as
> > bearer. No matter what type
there are diffs for both 1.2 & 1.4 - look at the links
On 12 January 2007 21:20, Goke Aruna wrote:
> Anton wrote:
> > Guys,
> >
> > Below is the link to a little modification for chan_ss7
> > allowing to address more than 8xSpans and utilizing ZAP
> > styl
Guys,
Below is the link to a little modification for chan_ss7
allowing to address more than 8xSpans and utilizing ZAP
style access to a zaptel devises, made by Denis Smirnov
http://download.seiros.ru/SeirosPBX/chan_ss7/
Regards,
Anton
hm, not sure I think i've seen it too.
try setting different hunting mode...
On 8 January 2007 13:53, [EMAIL PROTECTED] wrote:
> Did anybody has an idea?
>
> Thanks
>
> Nico
>
> On Wed, 3 Jan 2007, [EMAIL PROTECTED] wrote:
> > Hello,
> >
> > The Problem is now happening again all 2 to 3 days and
>
Matthew,
any progress with making a sync with 1.4?
Regards,
Anton.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-ss7
dpc => W05:8304,W03,8304
> links => l1:1,l2:2
>
> On Wed, 13 Dec 2006, Marc Storck wrote:
> > See the Thread: [asterisk-ss7] Some Feature-Questions
> >
> > It does not work for everybody, as the carrier on the
> > other side probably doesn't want or i
1 - 100 of 200 matches
Mail list logo
