elop against at the times that I have had open to work
on this.
Matthew Fredrickson
Digium, Inc.
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hat is not enough
to tell what is wrong.
Matthew Fredrickson
Digium, Inc.
>
>
>
> [May 18 11:19:08] ERROR[23463] chan_dahdi.c: Received MSU in invalid
> state 1
>
>
>
> Enabled debugging on linkset 1
>
> Link state change: NOTALIGNED -> IDLE
>
&
that you want added, that might be a good
motivation to fill it out :-)
Anyways, thanks again guys. I'll continue to keep everyone informed
about updates on the technical side as well.
Matthew Fredrickson
Digium, Inc.
--
_
code not match? Do you have an stp between?
Can you post a debug of the messages using "ss7 debug linkset x" where x
is the linkset number?
Matthew Fredrickson
Digium, Inc.
>>
>>
>>
>> -- Sent from my Palm Pre
>>
>> On May
What version of Asterisk are you using? This looks like the p->dialing
bug that some spoke of earlier, where it was not cleared properly. I
had thought that the fix got committed to all the relevant Asterisk
branches, but it's possible that maybe I missed one.
Matthew Fredrickson
Dig
nges back in, which is the other reason
why it has been so long and it still has not been merged.
If you're interested, either reply to me or this thread and let me know.
Thanks again,
Matthew Fredrickson
Digium, Inc.
--
__
IS (this is a
> NEC-specific protocol).
> And if it really TUP, how it will be difficult to implement this support?
I'm not sure, I haven't looked into TUP before, but if it's anything
like ISUP was, it's probably a non-trivial effort to do right.
Matthew Fredrickson
Digi
You'll have to turn on debug and get the whole message...
'ss7 set debug on linkset x' where x is the linkset number
Matthew Fredrickson
Digium, Inc.
Yoherman wrote:
> hi,
>
> anyone have experience with this :
>
> /Asterisk 1.6.2.0, Copyright (C) 1999
Lond_IP wrote:
> Hi!
>
> I have problem with ss7 link.
> Not in up mode.
The other end is sending you messages for TUP (user part 4), instead of
ISUP. libss7 right now only supports ISUP for bearer control, and not TUP.
Matthew Fredrickson
Digium, Inc.
>
> * 1.6.2.5 +
For Asterisk in released branches right now, the correct version of
libss7 to use is from the libss7 1.0 branch, or one of the 1.0.x release
tarballs of libss7.
http://svn.digium.com/svn/libss7/branches/1.0
Matthew Fredrickson
Digium, Inc.
Johann Steinwendtner wrote:
> Hello !
>
&
tarted for doing SS7
point code clustering... it uses a home grown protocol for message
transport over IP.
Matthew Fredrickson
Digium, Inc.
>
> Thanks
>
> Nfx
>
>
> > Date: Wed, 20 Jan 2010 14:03:27 -0600
> > From: cres...@digium.com
> > To: asterisk-ss7
this problem?
>
> thanks and sorry my English
Sorry, right now I'm not aware of a good sigtran solution for Asterisk
(other than buying an external gateway). At least as far as libss7 goes.
Matthew Fredrickson
Digium, Inc.
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>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-ss7
Can you provide the ss7 debug trace of the call (ss7 debug linkset x) in
order to verify that those parameters are ac
?
Actually, this sounds like a very good idea. I think that a better
indication might be a specific event related to this though, rather than
trying to overload abort events (since you can legitimately have abort
events on a link). I will think ab
Domjan Attila wrote:
> Hi,
> I wrote it here approx 2 weeks ago the missing 2 p->dialing = 0 lines in
> ss7 part.
Yeah, thanks so much for pointing that out. I had merged your change
into trunk, but the other branches had not done yet. Big thanks to
Domjan :-)
Matthew Fredrickson
it should contain the changes you need to fix it.
Please let me know if there are any remaining issues relating to this as
well if they come up.
Matthew Fredrickson
Digium, Inc.
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as
Freddi Hansen wrote:
> Hi,
> I am using libss7.
> I have TE420 with ss7 link to a Huawei m800 on one side and SIP on the
> other side of a gateway
I believe I know what is causing this problem. Can you contact me
directly about this via IM, and I will see if we can resolve
t;>
>>> Visual Audio Levels.
>>>
>>> Use chan_dahdi.conf file to adjust the gains if
>> needed.
>>> ( # = Audio Level * = Max Audio Hit )
>>> <----(RX)>
>>> ###*
rs only
31 physical channels, and people forget to add a new "cicbeginswith=33"
for the channels in the second span, causing an off by one CIC error
starting with the second span (and each subsequent span).
Matthew Fredrickson
Digium, Inc.
>
> Thanks,
>
> Daniel Pizarro
Daniel Pizarro Bustamante wrote:
> Hello Mesbah:
>
> I am using the libss7-1.0 because the libss7-trunk has an error that we could
> see in this forum.
>
> The asterisk and dahdi was downloaded from the svn (trunk versions).
>
> How can I detect the CIC mismatch?
Well, one way to do it is to
mistake.
So, to clarify:
If you need to have PRI/ISDN support, you must install libpri.
If you only need to have SS7 support, you can just install libss7.
Thanks,
Matthew Fredrickson
Digium, Inc.
> Thanks
> Mesbah
>
> On Sat, Apr 18, 2009 at 5:26 PM, Kashif Ali <mailto:k
ut I'm hoping they
are going to slow down relatively soon. When that happens, we'll start
merging your latest changes back into the mainline again.
Matthew Fredrickson
Digium, Inc.
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Sorry for the high latency on this, but it has been really busy. This
looks like it could be a bug in the configuration file parsing code.
Can you try the attached patch for me and tell me if the NAI goes to
what you set it to? (it should apply ok against 1.6)
Matthew Fredrickson
Digium
up_ubl’ makes pointer
> from integer without a cast
> chan_dahdi.c:15561: error: too many arguments to function ‘isup_ubl’
> chan_dahdi.c: In function ‘handle_ss7_unblock_linkset’:
> chan_dahdi.c:15611: warning: passing argument 2 of ‘isup_ubl’ makes pointer
> from integer without a cast
>
related to
high load related to realtime servicing of the link.
Matthew Fredrickson
Digium, Inc.
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o occur are found very
quickly since the dial plan is probably one of the most tested parts of
Asterisk :-)
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Matthew Fredrickson
Digium, Inc.
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le
wiggle room to change things around though.
---
Matthew Fredrickson
Digium, Inc.
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t; 1-15
> cicbeginswith=17
> channel => 17-31
> sigchan=16
Can you post a debug of this when it's occurring?
Also, can you post your chan_dahdi.conf as well?
Thanks,
Matthew Fredrickson
Digium, Inc.
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Those parameters (IIRC) will not be changed right now except on anything
but a reload of the entire module itself. Probably something I should
change :-)
To be safe, if you have a chance to restart Asterisk, please do so and
let me know if that fixes it. If it doesn't, it sounds like there m
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
you for your advice
The configuration in /etc/dahdi/system.conf only configures layers up to
MTP1/MTP2.
You set the ss7 signalling protocol selection (ISUP) in
/etc/asterisk/chan_dahdi.conf
Matthew Fredrickson
Digium, Inc.
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x27;re using the correct timeslot for signaling? (i.e.
what did they tell you to use)
Matthew Fredrickson
Digium, Inc.
>
>
> Here is my system.conf
> [r...@invpbx01 dahdi]# cat system.conf
> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 18 13:51:09 2009 --
> do not hand
nk branch that is separated from the 1.0 branch of
libss7, I will probably check the bug13495 changes (at least the libss7
part) into that branch soon. This branch will become 1.2 of libss7,
when the API has stabilized.
Matthew Fredrickson
Digium, Inc.
>
>> Regards,
>> At
quantity of trunks available
>>> under a single point code?
>>> (In the area of 28-500 T1s worth)
>>>
>>> An a-link could be made available to each box if necessary.
>
>
>> I'm actually working on this code right now... :-)
>
>> Matthew
Florian Smeets wrote:
> On 10.02.2009 18:34 Uhr, Matthew Fredrickson wrote:
>> Florian Smeets wrote:
>>> On 10.02.2009 17:59 Uhr, Matthew Fredrickson wrote:
>>>> Florian Smeets wrote:
>>>>> On 2/10/09 5:04 PM, Matthew Fredrickson wrote:
>>>>
Florian Smeets wrote:
> On 10.02.2009 17:59 Uhr, Matthew Fredrickson wrote:
>> Florian Smeets wrote:
>>> On 2/10/09 5:04 PM, Matthew Fredrickson wrote:
>>>> Florian Smeets wrote:
>>>>> These are messages from the switch:
>>>>>
>>
Florian Smeets wrote:
> On 2/10/09 5:04 PM, Matthew Fredrickson wrote:
>> Florian Smeets wrote:
>>> These are messages from the switch:
>>>
>>> Msg:(E_SS7SS_MTP3_SLTM_FAILED) - SLTM failed. [Link : 160, OPC : 0x3e8,
>>> DPC : 0x708], Cause: SLTA_NOT_RE
Florian Smeets wrote:
> On 1/27/09 6:12 PM, Matthew Fredrickson wrote:
>> Florian Smeets wrote:
>>> Hi
>>>
>>> i'm trying to connect a Motorola MSC to asterisk-1.6.1-beta4 + libss7 trunk.
>>>
>>> I'm using FreeBSD and dahdi-bsd, i kn
Matthew Fredrickson wrote:
> Markus A. Wipfler wrote:
>> Hi Group,
>>
>> I am using libss7 to interconnect with telecom. I need to set the
>> Nature of Address Indicator depending on where the call is coming from
>> (SIP). How can i do this ?
>
> Ooh..
of the call when you
cannot do this...?
Maybe some SS7 debug output too (since this *is* the -ss7 list) :-)
Matthew Fredrickson
Digium, Inc.
>
>
> Any Idea?
>
> Olivier
>
>
>
>
>
> Dahdi config
>
> span=1,0,0,ccs,hdb3
> bchan=1-15
> mtp2=16
>
ange the source code.
Right now, we have a fixed NAI parameter that is set inside chan_dahdi.conf.
You might be able to add a simple dialplan variable to set it though.
There are plenty of good examples of doing things like this with other
SS7 variables in chan_dahdi.c
Matthew Fre
re at work.
Can you post some debug output of this when it is occurring?
Thanks,
Matthew Fredrickson
Digium, Inc.
>
>
>
> And the telco sees us still as looked.
>
>
>
> On our server, we have 3 digium cards. For what I understand from the
> telco,
>
> we
terisk does not speak
sigtran, only traditional SS7 at this time :-(
Matthew Fredrickson
Digium, Inc.
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alent of /etc/dahdi/system.conf have in it?
The real thing that would tell us a lot is to ask the switch at the
other end why it's dropping the link.
So far, it looks like all the messages are good, point codes are
correct, etc.
Matthew Fredrickson
Digium, Inc.
__
for catching this. In trunk, they have been redoing how the CLI
commands work, so it looks like in whatever revision you checked out,
you got one with a CLI inversion bug :-)
Please try to update, and let me know if it is still there.
Thanks,
Matthew Fredrickson
Digium, Inc.
Mehdi Shirazi wrote:
>>> Although I am not using true M3UA and sigtran protocols right now,
>>> conceptually, this is almost exactly what we are doing, except we will
>>> support multiple SGs to each ASP as well.
>
>>> Matthew Fredrickson
>>> Digium
Whats your opinion guys?
I have not tried to add MGCP support to chan_zap/chan_dahdi... Making a
channel driver (probably a new one) that talks SS7 and MGCP though is
next on the list after I'm pretty happy with the way clustering is
working though.
Matthew Fredrickson
Digium, Inc.
Ruddy Gbaguidi wrote:
> Hi All
> I have to setup an ss7 link with the telco but they are not seeing any
> signal from my side.
>
> Behind the digium card the light is GREEN however. Is there anything I'm
> missing here ?
> Here is are my configuration.
You can also try using the ss7linktest pro
\ /
>
Although I am not using true M3UA and sigtran protocols right now,
conceptually, this is almost exactly what we are doing, except we will
support multiple SGs to each ASP as well.
Matthew Fredrickson
D
.
I think I agree with that. It would definitely make things simpler when
doing ISUP masquerading... (as long as we can count on other switches to
not cross T1/E1 boundaries).
Matthew Fredrickson
Digium, Inc.
>
>> For A links, I think it won't be a problem though because the CIC
o trunk anyways,
I just haven't done it yet, so eventually they'll be in trunk.
http://svn.digium.com/svn/libss7/branches/mattf/bug13495
http://svn.digium.com/svn/asterisk/branches/mattf/bug13495
Matthew Fredrickson
Digium, Inc.
>
> Regards,
> Attila
>
>> Well, to
Amish Chana wrote:
> Matthew Fredrickson wrote:
>> LES.NET (1996) INC. wrote:
>>
>>> Hello.
>>>
>>> I'm curious if anyone has had asterisk-ss7 working (through whatever
>>> magic) across multiple boxes under a single point code.
>>&g
> An a-link could be made available to each box if necessary.
I'm actually working on this code right now... :-)
Matthew Fredrickson
Digium, Inc.
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debug to trigger based on that information since debug begins
indiscriminately at the MTP2 level and ripples up to MTP3, and then ISUP.
Matthew Fredrickson
Digium, Inc.
>
> It could be very useful, once we can check call by call separately
>
> For example
>
> ss7 debug linkset
just set the SOFTHANGUP flag on the
ast_channel so that Asterisk will initiate the hangup at that point.
That is how it is done in libpri in a similar scenario, if you look at
PRI_EVENT_PROGRESS handling code in chan_dahdi.c. (IIRC)
Matthew Fredrickson
Digium, Inc.
>
> On Thu, 2008
re you don't need CRC on the line) is to try to check
your cabling and replace it if possible.
Matthew Fredrickson
Digium, Inc.
>
> /etc/dahdi/system.conf
>
> span=1,1,0,ccs,hdb3
> bchan=1-15,17-31
> #dchan=16
> mtp2=16
>
> /etc/asterisk/chan_dahdi.conf
>
he's going to try to hard code it to
"Charge" instead, to see if it fixes his issue. If it does, we might
add a configuration parameter for this.
Matthew Fredrickson
Digium, Inc.
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ome Monday or so, it's probably a misconfiguration
> somewhere.
That's what it sounds like to me as well. Can you configure the PRI and
have it come up separately, without the SS7 link?
Matthew Fredrickson
Digium, Inc.
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without modifying source cods ?
> If need modifying source code is it easy or need deep understanding of
> source codes ?
> what is your suggestion is it practical idea at all or not ?
This is certainly doable and a not uncommon situation. I think a nu
ike you
have done) and it will work.
Matthew Fredrickson
Digium, Inc.
>
> group=0
> relaxdtmf=yes
> signalling=ss7
> ss7type=itu
> ss7_calling_nai=national
> ss7_called_nai=national
> linkset=1
> pointcode=10567
> adjpointcode=10619
> defaultdpc=10581
> networ
made
the copy from. I'll see if I can update the branch sometime today or
monday to see if it is fixed.
Matthew Fredrickson
Digium, Inc.
>
>
>> http://svn.digium.com/svn/asterisk/team/mattf/bug13495
>>
>> ---
>> Matthew Fredrickson
>> Digium, Inc.
&
Anton wrote:
> 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
http://svn.digium.com/svn/libss7/team/mattf/bug13495
http://svn.digium.com/svn/asterisk/team/mattf/bug13495
---
Matthew Fredrickson
Digium, Inc.
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Rana Dhekial wrote:
> Hi Matthew,
>
> As per Telco both these SS7 links are hooked to the same adjacent point
> code. I will confirm from telco. SS7 debugs are as follows
Why are you getting red alarm on those spans?
That could be part of the problem.
Matthew Fredrickson
. Can you check if I
> have configured it wrong ?
Can you post an ss7 debug on the linkset which includes the link that's
fluctuating?
Also, it looks like your adjpointcode is the same for both links. Is
this correct? (both links being hooked up to the same point code at the
other end).
t seems that this would be a clear way to
automatically hang up in this case.
Matthew Fredrickson
Digium, Inc.
>
>
> <>
> -- Executing [EMAIL PROTECTED]:1] Macro("SIP/sky_ktm01
> -08c167e0", "trunkdial,DAHDI/g2/9851060166") in new
> How can i void sending ANM messages and/or release the call with a
> different cause code when the there is some errors on the
> applicationsl?
To set the cause code on the outbound REL, you can set the SS7_CAUSE
channel variable on the channel you are hanging up.
Matthew Fredrickson
interconnection, the another
> telco has STPs the current not, so the STP related stuff not tested yet
> with switch (inhibit, TFP).
> I will do some more tests tomorrow and I think after it should merging
> the patches.
Great to hear!
Matthew Fredrickson
Digium, Inc.
>
> Regard
ing in LIBSS7?
There shouldn't be any issues. Can you use dahdi-monitor to verify that
the tones are not actually being sent out on the line when you think
they should be?
If they aren't please contact me so we can get this fixed.
Matthew Fredrickson
Digium, Inc.
>
> Setup:
&g
Tobias Wolf wrote:
> Matthew Fredrickson schrieb:
>> Kristian Nielsen wrote:
>>> Tobias Wolf <[EMAIL PROTECTED]> writes:
>>>
>>> I believe you can do this with chan_ss7, setting up two boxes as a
>>> cluster, though I think you would set
ultiple machines.
> possible to do it using two different point codes as you describe with some
> help from the switch at the other end?
That's the way I would advise doing it for the time being...
Matthew Fredrickson
Digium, Inc.
>
> - Kristian.
>
>> is the following
Anton wrote:
I can go to IRC when needed. Just tell me when.
Regards,
Here's a patch. Apply it to your asterisk source tree.
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
r anything like that and can get in contact with
me, I would like to fix this (should be just about 5 minutes of coding
and a retest from your test scenario).
Matthew Fredrickson
Digium, Inc.
>
> It means, when there are some amount of calls going through the system, and
> remote switc
Krzysztof Drewicz wrote:
> 2008/11/3 Matthew Fredrickson <[EMAIL PROTECTED]>:
>> Krzysztof Drewicz wrote:
>>> 2008/11/3 Matthew Fredrickson <[EMAIL PROTECTED]>:
>>>
>>>> Is this using the signaling channel in /etc/dahdi/system.conf as a dchan
Krzysztof Drewicz wrote:
> 2008/11/3 Matthew Fredrickson <[EMAIL PROTECTED]>:
>
>> Is this using the signaling channel in /etc/dahdi/system.conf as a dchan
>> or as mtp2?
If EC is indeed disabled, it's entirely possible that you're running
into your CPU limit
Krzysztof Drewicz wrote:
> 2008/11/3 Matthew Fredrickson <[EMAIL PROTECTED]>:
>> Well, the only thing that might cause the CPU to not be able to handle
>> packets properly with the mtp2 interface is software echo cancellation
>> (since it also happens at the same priori
Krzysztof Drewicz wrote:
> 2008/11/3 Matthew Fredrickson <[EMAIL PROTECTED]>:
>
>
>> I highly doubt that's your problem. You could run an SS7 link on a
>> really low end processor and still be able to handle packets, at least
>> using the Zaptel/DAHDI MTP2
ur problem. You could run an SS7 link on a
really low end processor and still be able to handle packets, at least
using the Zaptel/DAHDI MTP2 interface.
It looks like you're getting alarms on the link. You should fix that first.
Matthew Fredrickson
Digium, Inc.
> i set:
> mtp2=
Krzysztof Drewicz wrote:
> 2008/10/10 Matthew Fredrickson <[EMAIL PROTECTED]>:
>
>>> Hi, i want to know how to set a SS7 without sigchan. My SS7 Provider
>>> said that the SS7 do not have signalling channel because is a
>>> enlargement voice SS7.
>
; configuration, it is so strange for me.
There is no SS7 without a signalling channel. How are you going to
signal calls without getting any signalling about which channels have
calls on them?
Matthew Fredrickson
Digium, Inc.
>
> This is my ss7 /etc/dahdi/system.conf, i have i
di_ss7_error: !!
> Unable to handle message of type 0xd on CIC 11
Well, this is the suspend message. We currently don't do anything
interesting with it, but it should be benign at the very least. There's
actually a bug on the bugtracker adding more full support for it though.
Hope
mtp2 channel type in Zaptel/DAHDI (if you're using a Digium board).
Matthew Fredrickson
Digium, Inc.
>
>
>
> On Sep 23, 2008, at 9:38 PM, Matthew Fredrickson wrote:
>
>> Markus A. Wipfler wrote:
>>> Hi all,
>>>
>>> I am getting below e
ils.
Can you describe the problem (more exactly) that you are having better?
(other than the SCCP part)
Thanks,
Matthew Fredrickson
Digium, Inc.
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To UNSU
always just remove them by hand. You might try
`make uninstall` though in the respective directories (I *think* there
might be an uninstall target).
What is the current problem you are having right now, so that we can
better help you perhaps?
Matthew Fredrickson
Digium, Inc.
___
lc in the Makefile (basically
remove all references to it there).
Matthew Fredrickson
Digium, Inc.
>
> /tom c.
>
>
> - Original Message -
> *From:* olivier taylor <mailto:[EMAIL PROTECTED]>
> *To:* asterisk-ss7@lists.digium.com
> <mailto:
in another email that there is no echo anymore for you:
"Hi Matthew,
No problem, it works, problem was coming from my provider.
Regards,
Olivier"
Matthew Fredrickson
>
> Kind regards,
>
> Olivier
>
>
>
> Matthew Fredrickson a écrit :
>> olivier ta
cluding the SS7 support in
chan_dahdi).
You can certainly make that box the IVR and the PBX, although it's a
configuration you'll need to test to make sure you're happy with it.
Matthew Fredrickson
Digium, Inc.
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Attila Domjan wrote:
> + 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
Thanks. I'll start looking at this when I'm done reviewing another patch.
Mat
last time when i posted the message :
That's an entirely different issue then if the span's in alarm. You
need to figure out why it's in alarm then if that's the case. Always
fix the lowest level
what do you have in you
/etc/dahdi/system.conf?
Matthew Fredrickson
Digium, Inc.
>
> Kind regards,
> Olivier
>
> Matthew Fredrickson a écrit :
>> Sriram wrote:
>>
>>> Hi
>>>
>>> output of ss7linktest :
>>>
>>> Could n
seeing
either SIO or SIOS received ('<'). It is likely that the other end has
not turned the link up. Just tell them you're not seeing SIO or SIOS
from them on the link when you try to align the link.
Matthew Fredrickson
Digium, Inc.
__
Sriram wrote:
> Hi
>
> output of ss7linktest :
>
> Could not open device 16 : Device or resource busy
You can't run ss7linktest unless you shut down asterisk first. This is
probably why you're getting Device or resource busy.
Matthew Fredrickson
Digium, Inc.
>
to the modem everything is right
Can you try running ss7linktest on your link and give the output of
that? That will tell what we see from the very beginning on that link.
Matthew Fredrickson
Digium, Inc.
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ponding to our
request to align the link (by sending SIO).
If you have a TE2xxp card (TE210 TE205 or TE220) you should update your
DAHDI version to latest trunk as there was a bug in that driver which
would have caused an issue similar to this. Otherwise, you might check
with you
nkset (although now we do
support this), so it's looking through all the CICs on the linkset and
every time it finds the CIC specified, it blocks it (regardless of DPC).
It's something I"m going to have to fix...
Matthew Fredrickson
Digium, Inc.
> it will block me 2 cics :
fore...
>
> Same here,
> dahdi make terrible echo...
Did you configure an echo canceller in /etc/dahdi/system.conf?
If so, make sure you're using MG2 (the default for Zaptel)
Matthew Fredrickson
Digium, Inc.
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ed alarm on channel 1:Red Alarm --registered channel
> 1, ss7 signalling
> /this continues till channel 5 now and exits saying cant set signalling
> on channel 6. / /
Try moving your sigchan= line above your channel= line in the linkset
configuration.
Matthew Fredrickson
Digiu
vvc) and tell me why chan_dahdi is failing with the
increased verbosity.
Matthew Fredrickson
Digium, Inc.
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asterisk-ss7 mailing list
To UNSUBSCRIBE or update options v
mum so far though.
Matthew Fredrickson
Digium, Inc.
>
> Regards,
> Attila
>
> On Sat, 2008-09-13 at 12:39 -0500, Matthew Fredrickson wrote:
>> In order for me to review these to be included into libss7 and
>> chan_dahdi, you will need to submit them via bugs.digium.co
chan
> on channel 16the operator is still doing some chnages at his end but
> i just want to confirm if this is some issue at his end or my end
>
> thanks- sriram
That is an issue with your end. Make sure you signalling is set to
either mtp2 (if it's a Digium card) or dch
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