a bug and someone
requested a patch but nothing lately.
Any clues would be very helpful.
Thanks
Joseph
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We use allo.com cards and they are software configured for 56k in the dahdi
config files.
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of code0071
Sent: Wednesday, April 15, 2015 10:08 PM
To: asterisk-ss7@lists.digium.com
Subject: [asteri
as correct? This
doesn't appear to be a CIC problem yet per se - you're not able to get
the signaling link to come into alignment.
Matthew Fredrickson
On Fri, Nov 7, 2014 at 1:07 PM, Joseph Jackson wrote:
> Hey list,
>
>
>
> So I’m trying to get ss7 turned up between an
the linkset reports as up tho I still see those
NOTALIGNED -> IDLE messages over and over.
Customer hasn’t reported back yet what their end see’s so I’m waiting on that.
Joseph
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asterisk-ss7-boun...@lists.digium.com] On Behalf Of Vi
ith=1
channel=1-23
channel=25-48
adjpointcode=XXX-XXX-XXX
sigchan=24
Any help would be very appreciated.
Thanks!
Joseph Jackson
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I have changed to:
; local interconnect
cicbeginswith = 2
sigchan = 1
channel = 2-16
;international interconnect
group = 2
ss7_calling_nai=international
cicbeginswith = 17
channel = 17-31
Still getting "Network Indicator: 3"
Joseph
On 16 May 2014 11:44, peterpet wrote:
>
= 2-16
;international interconnect
group = 2
ss7_calling_nai=international (I have also tried dynamic here with same results)
channel =17-31
If I change the first ss7_calling_nai to dynamic, all calls go with
Network Indicator: 4
Joseph
On 15 May 2014 10:35, Marcelo Pacheco wrote:
> Yes, m
ss7_calling_nai as national for CIC 1-16 and
ss7_calling_nai as international on the same E1?
Joseph Mpora
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Does this help or work?
http://www.voip-info.org/wiki/index.php?page=Asterisk+libss7#CallRedirection
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regards, Joseph
On Nov 9, 2011, at 2:54 PM, Sal Aguilar wrote:
> Hi,
>
> Anyone know if there is a fork of libss7 that supports Redirecting according
> to ISUP'92? (
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smime.p7s
Description: S/MIME cryptographic
On Mon, Aug 23, 2010 at 11:00 AM, joseph mpora wrote:
> Hullo all,
>
> I have successfully setup dahdi + libss7 + asterisk and call traffic
> seems normal.
>
> There is just 2 issues I need help resolving:
>
> 1. I get the following warning on asterisk cli very often: &q
hould have 2 signalling links each on the 1st channel of the 1st
and 2nd E1. The telco sees only one link up and ss7 show linkset 1
only show one link. How do I configure/enable the other?
Thanks, Joseph
Key Information:
Asterisk 1.6.2.9
libss7 version: 1.0.2
DAHDI Version: 2.3.0.1 Echo Canc
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Joseph - t...@ekn.com
IT Manager / Appalachian Wireless
East Kentucky Network, LLC.
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Thanks.
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th - but
> hardcoding this to 1, and then setting screening indicator correctly allow
> the function to work.
>
> how do I report a bug ?
File a report at https://issues.asterisk.org
If you can include a patch, that would be great as well.
Thanks.
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_
bearing data.
> Can I do this with Libss7? Or need to use Chan_ss?
> If I can do with Libss7, I need some clue,hint, apointment... to keep
> looking for.
> Thanks to all
There is no stable released libss7 to do this. You can try chan_ss7,
but that is not supported by Digium.
Than
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Are you able to test with A links?
On Jul 20, 2009, at 10:58 AM, Girish Agarwal wrote:
> Joseph,
> I am using Different Signalling Channel(1)and Voice
> Channels(2-24). Nortel side is the Clocking source for the T
exten => _19549776740,1,VoiceMail(${CALLERID(dnid)}...@default)
> >value 19549996740
> ;exten = _19549993738,1,VoiceMail(${CALLERID(num)}...@default)
> >value 19549996740
> ;exten = _19549776740,1,VoiceMailMain(${CALLERID(num)}...@default)
> >value 19549996740
>
ips on how to prevent that from happening? Does the
order of the T1 setup have any impact on this?
Like have the A links first or last in the listing...
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regards, Joseph
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Version: GnuPG v1.4.9 (Darwin)
iEYEARECAAYFAkpnCZUACgkQ5CyZqOno04yd6wCeOxc2vm0bwytj+s
back from Asterisk has pointcode as DPC(05-00-00) OPC (05-00-00 )
> which looks like a bug in the system.
>
>Relevant Portion of chan_dahdi.conf file is as follows:-
>
> ss7type = ansi
> linkset = 1
> pointcode = 5-15-93
> adjpointcode = 5-15-80
> networkindicator=
in entirety. I've also tried pulling
> down versions of Asterisk all the way back to asterisk-1.6.0.1.
>
> Any suggestions
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Version: GnuPG v1.4.9 (Darwin)
iEYEARECAAYFAkpcjPIACgkQ5CyZqOno04zyNQCgh7yi7ldVx1Eybh5ALIXxRaQL
YLQAni8ruEqK92O
mens EWSD-ITU C7--->Asterisk
> EuroISDN->Cisco AS5300/Sysmaster SM7000
>
Yes, this should work fine.
You just need to setup your hardware and configure it appropriately.
- --
respectfully,
Joseph - t...@ekn.com
-BEGIN PGP SIGNATUR
n't see ss7 related cli
> commands like
> ss7 show linkset x
> ss7 debug linkset
>
> And there is no mention of ss7 on asterisk-1.4.22 source code.
>
> Does 1.4 support ss7 or I really needs to install asterisk 1.6 ??
You will need to go with 1.6.
--
respectfully,
Josep
On 12/04/08, Joseph wrote:
> On 12/03/08, Anton wrote:
> >
> > http://svn.digium.com/svn/libss7/team/mattf/bug13495
> >
> > http://svn.digium.com/svn/asterisk/team/mattf/bug13495
> >
I have tested this and was not able to use it.
In my test, I am using ansi w
tch type: ANSI
Our point code: XX
However that point code does not make sense since I am using ansi.
It looks like hex or something.
Can we have that formated like xx-xx-xxx ?
Thanks.
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Joseph
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This means there is a problem with the ss7 link.
It might a speed issue, like 56k instead of 64k or vice versa.
Check the debug on the far end if you have access to it.
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Joseph
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IC from 33 to 49 also no
> singalling link (SLC=1) on 48 D channel.
> anyone has got it working?
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Joseph
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a
On 09/02/08, Rony Ron wrote:
> Hi Joseph,
> your solution is very elegant,
> what are those parameters:
>
> _SS7_LSPI_IDENT=ON
> _SS7_RLT_ON=YES
>
This tells * to drop out of the call path and let the ss7 provider take
it back.
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Joseph
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k you logs etc and see what is failing.
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Joseph
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ansfer' is not really
> working and you are using two channels for one call..
> this is not only related to -ss7, but also to -zap digital channels.
>
> look for 2bct, or ECN or cmd Transfer() in voip-info.org
>
Libss7 has limited redirect support with ss7.
It has be
On 09/02/08, Virmones Pereira wrote:
> Regards for everybody
>
> Joseph your solution is very interesting, but I think I will have another
> two problems the first trouble is with billing, Because if I answer the call
> it will be charged right?
> And the second trouble is when
>
> channel = 1-15,17-31
>
> defaultdpc = 5
> channel = 32-46,48-62
Yes, that looks good.
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Joseph
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using asterisk as a STP where the SS7 use
> > only the signaling channel, the media should go directly to the SSP
> >
> > somebody knows how to do it?
> >
> >
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Joseph
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s one linkset with 2 ss7 links.
So, 1st signalling channel could be channel 16 and go to SW1, 2nd signalling
channel
could be 17 and go to SW2.
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Joseph
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On 08/27/08, Grant Arix wrote:
> Hi all,
>
> Have you ever tried Libss7 with Linux x86_64 ?? especially with FC 9 x86_64 ?
It runs fine on x86_64 and Debian. I don't know about FC... but expect
it should be fine.
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Description: Dig
ror: ‘bi’ undeclared (first use in this function)
> make: *** [ss7linktest] Error 1
>
> Any idea of what's wrong?
You need to use dahdi instead of zaptel.
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Joseph
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gt;
It has worked for others.
What you might check is the bchan speed.
It is possible that you will need to use d4 and ami which is a 56k type
channeling.
--
respectfully,
-
|Joseph |
-
f you can test and post your results, that would be great.
Obviously since the option is there someone has worked with 56k
linksets.
--
+
respectfully, Joseph |
+
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Description: Dig
oks like you would need to modify libss7 to not send the IAM until it
got a Dial/Answer or something that would make the channel open.
You might start by looking at the libss7 stuff.
Grep for IAM to get started.
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t a 64k link.
Thanks.
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look at your Dial options. "show application dial"
I think adding the m will tell it to play music.
exten => _07892111XXX,1,Dial(Zap/37/0${EXTEN},60,m)
--
+
respectfully, Joseph |
+
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On 04/24/08, Barry O'Donovan wrote:
>
> Anton, Paul, Joseph,
>
> Thanks for your assistance and suggestions to date.
>
> Based on the information you provided, I think I may have narrowed the source
> ot the problem down the the timing source.
>
> I have my
le zaptel.conf and zapata.conf settings included
with asterisk 1.6 and zaptel 1.4?
If you have red or yellow alarms, you have something wrong with sangoma
or zaptel.conf settings.
--
+
respectfully, Joseph |
--
the settings you used with libss7?
Also do you know what speed the A/F link is that you are connecting to?
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though if you need it :)
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ht
h more than one linkset. But mixing dchan and
bchan types has not worked to well. I would suggest moving the dchan to
the end of the bchan.
If it works with one linkset but fails when adding the second, than you
know there is a configuration problem.
--
+ respectfully, Joseph =+
Axe10 gateway?
As I understand, they are ss7 messages that have not yet been defined in
libss7. You are welcome to submit patches to get these settings in if
you like :)
Likely they do not have any performance impact, however, extra features
that they relate to may not be available
and ansi
;ss7type = itu
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asterisk
Dial(zap/${EXTEN}
It will be like any other Zap channel.
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ard with "cat /proc/zap*"
And make sure you and your provider both are doing either 56k or 64k.
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_
see if that
works.
This is possibly a bug as I got around it by moving the sigchan channels
to the end of the card with wiring etc and it works.
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-
signature
rds
> Ayman
Yes.
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On 03/30/08, Raymond Jender wrote:
>
> Anyone experimenting with 1.6? I'm sort of new at
> this. If you're using 1.6 are you using a GUI?
No gui, just vi (vim) :)
--
respectfully,
-
|
the other E1, this keeps happening
> randomly. I'm testing with ss7box.
I don't know about ss7box.
But it sounds like a CIC miss match.
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sig
his
"asterisk -cddg"
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On 03/22/08, Nitesh Divecha wrote:
> Joseph,
>
> Thanks man for the info...
>
> Yea, I am going to setup a separate box but is it possible to use SIP? I
> am planning to route the calls directly to the SS7 Asterisk box from my
> softswitch. The softswitch we are using on
ndancy today is with multiple
Point Codes and separate boxes.
The other stack is chan_ss7 which someone who has used it can help you
with.
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Descriptio
e isup.conf file is part of chan_ss7, not libss7.
Thanks.
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like the only working redundant setup requires multiple point
codes with multiple complete hardware configs, and fail over routes in the
switch/MSC.
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-----
| Joseph | IT |
--
it is exist
This might be a question to ask on the normal asterisk-users list.
Here we are mainly looking * and ss7 integration.
I am sure what you want to do is possible.
--
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-----
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---
.
This only applies to the channel/ds0 you use for ss7.
Google should get you a bit more info if you need it.
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as
f and
zapata.conf. The options are listed in the sample configs included with
asterisk head.
Thanks.
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Description:
roviders?
Try blocking one and have them see if the same cic was blocked.
To block one do "ss7 block cic 1 19" This will block linkset 1 cic 19.
The provider should see the block. You can use unblock to reverse the
block.
--
respectfully,
-
|
On 02/09/08, Everton Goularth wrote:
> Thank's Joseph for the suggestion,
>
> but I have already looked at both (zaptel and zapata) samples config files.
>
> As a mater of fact, my dchan for my first E1 is the channel 16, and because
> this I'm
> think that i
sample configuration.
This seems to strongly indicate zapata.conf or zaptel.conf configuration
error. Perhaps sigchan 16 is not a dchan in zaptel.conf.
--
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| respectfull, Joseph |
signatur
ne is the real MSC.
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asteris
of the debugging which shows that my box is
> sending REL. I don't really know what is the reason for sending this
> message.
>
First, you need to confirm that you don't have a dial plan issue where
the call cannot be completed.
I had the same problem, and it was a
se seen this?
Any chance you can try it on 64 bit?
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unk asterisk
ln -s zaptel1.4 zaptel
>
> I don't know if is possible to do , can you help me?
>
I am glad to try :)
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sk, libpri, libss7 to all come from head.
And then zaptel to come from 1.4
Start with a clean setup and try again, as it works here.
--
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+
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want to make defaultdpc point to the PC that controls your cic
trunks. Adjpointcode simply means what the next hop is to get to defaultdpc.
I have seen it work even if the adjpointcode is wrong. I am not sure how
that can be...
--
--
EST
> [ 81 ]
> OPC 1001 DPC 100 SLS 0
> [ 64 40 fa 00 ]
> [ 21 ]
> H0: 1 H1: 2
> Message type: COA
>
> Len = 9 [ 83 82 06 80 64 40 fa 00 17 ]
> FSN
Matthew Fredrickson wrote:
> Joseph wrote:
>> Matthew Fredrickson wrote:
>>> Joseph wrote:
>>>> Matthew Fredrickson wrote:
>>>>> Joseph wrote:
>>>>>> Matthew Fredrickson wrote:
>>>>>>> Joseph wrote:
>>>
Matthew Fredrickson wrote:
> Joseph wrote:
>> Matthew Fredrickson wrote:
>>> Joseph wrote:
>>>> Matthew Fredrickson wrote:
>>>>> Joseph wrote:
>>>>>> Is it possible to get ss7 working using an A-link?
>>>>>>
>>&
Matthew Fredrickson wrote:
> Joseph wrote:
>> Matthew Fredrickson wrote:
>>> Joseph wrote:
>>>> Is it possible to get ss7 working using an A-link?
>>>>
>>>> Could you put the alink on one t1 time slot and then bring bearer
>>>> tr
Matthew Fredrickson wrote:
> Joseph wrote:
>> Is it possible to get ss7 working using an A-link?
>>
>> Could you put the alink on one t1 time slot and then bring bearer
>> traffic in on the 2nd, 3rd and 4th t1s?
>
> Yes, you definitely can do this in libss7, as l
Is it possible to get ss7 working using an A-link?
Could you put the alink on one t1 time slot and then bring bearer
traffic in on the 2nd, 3rd and 4th t1s?
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