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Marek Cervenka
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possibility to make it works?
In which ss7 parameter i can get Original Called Number in asterisk?
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Marek Cervenka
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but I cant see anything in
SIP, is there a way to get the Generic Number through chan_ss7?
Regards,
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Marek Cervenka
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the problem is incomplete calling party number
check the "Address signal" attribute
sangoma fills the X-Freetdm-GN parameter
http://wiki.sangoma.com/ftmod-sangoma-ss7-generic-number
SS7 IAM
-|--|
DPC to share signaling
links. More information is available athttp://www.netfors.com/chan_ss7
Great news!
thank you Anders
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Marek Cervenka
merge this into digium's libSS7 ? (are you willing to
sign disclaimer for digium?)
chan_ss7 is not developed anymore from your move from sifira to mysql i
think
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Marek Cervenka
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Marek Cervenka
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hi,
do you have someone:
- libSS7 in production use? how many concurrent channels?
- ported chan_ss7 to 1.4.13 and in production use? how many concurrent
channels?
thanks
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Marek Cervenka
ocal/remote hangup info in NOTICE (cervajs at freevoice.cz)
please test and report
thanks
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Marek Cervenka
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ches
> (ZTE and HUAWEI for me) it works without this change fine.
>
> On Saturday 17 November 2007, marek cervenka wrote:
>> hi,
>>
>> i made tarball with some ss7 patches from
>> www.voip-info.org and other places and put this at
>> http://www.freevoice.c
rom incompatible pointer type
> l4isup.c: In function `ss7_new':
> l4isup.c:569: error: too many arguments to function `ast_channel_alloc'
> make: *** [l4isup.o] Error 1
>
>
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Marek Cervenka
===
_
z/chan_ss7/chan_ss7-0.10.1.tar.gz
md5sum a3ca3031f8f4ef96d505be3b297b47cc
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Marek Cervenka
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asterisk-s
ls => 2-31
schannel => 1
firstcic => 1
enabled => yes
echocancel => 31speech
echocan_train => 350
echocan_taps => 128
[host-ss7box]
enabled => yes
opc => 0x1001
dpc => siuc:0x1000
links =
15000,timeout
> subservice => auto
>
> [link-l1]
> linkset => siuc
> channels => 2-31
> schannel => 1
> firstcic => 1
> enabled => yes
> echocancel => 31speech
> e
using sangoma a104dx on both
>> machine.
>>
>> I followed the write up on
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7
>> +setup
>>
>> I have the cross over cable between them.
>>
>> however, wanpipe shows connected but the signal
OH
>> TE_SIG_MODE = CCS
>> FE_TXTRISTATE = NO
>> MTU = 1500
>> UDPPORT = 9000
>> TTL = 255
>> IGNORE_FRONT_END = NO
>> TDMV_SPAN = 1
>> TDMV_DCHAN = 0
>&
gt; AUTO_PCISLOT= NO
>>> PCISLOT = 4
>>> PCIBUS = 15
>>> FE_MEDIA= E1
>>> FE_LCODE= HDB3
>>> FE_FRAME= NCRC4
>>> FE_LINE = 1
>>> TE_CLOCK= NORMAL
>>>
s7-1.0.9.tar.gz
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Marek Cervenka
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ou made my day,
Antoine Megalla.
- Original Message -----
From: "marek cervenka" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, February 20, 2008 6:16 PM
Subject: Re: [asterisk-ss7] "No Idle Circuit found"
after sometime
On Wed, 20 Feb 2008, Antoine Megalla 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
list)
tnx
---
Marek Cervenka
use
centos5(rhel5) is used for www.trixbox.org -> good tested
> Also, as I see, we have to possibilities about SS7, Chanss7 and LibSS7,
> wich is the best, easier,
libSS7 is in development stage, but digium supported
chan_ss7 is for production use (supported by www.dicea.dk)
--
e conversation needs a jb.
yes. if you terminate from SIP(outgoing call to PSTN) you need jb at
chan_ss7 side
PSTN <---(chan_ss7 w/jb) Asterisk SS7 <SIP SIP phone
in reverse direction is jb in the phone
PSTN --->(chan_ss7) Asterisk SS7 SIP> (jb) SIP phone
---
m a channel
> technology which causes jitter or not?
because of
Type: SIP (in channel)
(in one call are 2 channels bridged)
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Marek Cervenka
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ng again on link 'l1'.
do you have L1 up?
send us "wanrouter status"
>>> I am having problm chan_ss7 and sangoma 108d card, can any one help
>>> me. My telco is using Time Slot 1 as a singnal channel and 16 th
>>> time slot i
ilisé parce l'édition de lien n'a pas été faite
gcc: /usr/include/asterisk/: fichier d'entrée d'édition de liens n'est pas
utilisé parce l'édition de lien n'a pas été faite
cc1: /usr/include/zapte
process_event: MTP is now DOWN on link 'l1'.
>> [Aug 6 11:07:43] WARNING[25980]: mtp.c:1931
>> mtp_thread_main: No signalling links inservice and no
>> cluster receivers alive, dropping packet!
>> [Aug 6 11:07:45] NOTICE[25980]: mtp.c:1138
>> mtp2_good_fram
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> I experience this issue since 0.8x ... switched to libss7 because of it.
it was dtmf issue fixed in chan_ss7 1.1
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Marek Cervenka
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ou want scale over. then you need mtp3d (cluster solution)
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w/Asterisk+libss7 ?
thanks
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> schannel => 16
> firstcic => 1
> enabled => yes
> echocancel => no
> echocan_train => 350
> echocan_taps => 128
>
> [host-sifon]
> enabled => yes
> opc => 1234
> dpc => mylinkset:4321
> links => l1:1
> ssn => 7
>
> [jitter]
>
>
> Any comment is welcome.
rent i'm running chan_ss7 from svn trunk (rev 24), which seem to
> fix some compile issues with DAHDI.
> But if you are sure, I can try again zaptel. Must I downgrade from asterisk
> 1.4 to 1.2 again? Or asterisk-1.4 and zap
linkset => ss7trunk
channels => 2-31
schannel => 1
firstcic => 1
echocancel => allways
[host-some.host]
enabled => yes
links => l1:1
opc => 0x
dpc => ss7trunk:0x
[jitter]
jbenable = yes
jbmaxsize = 1000
jbresyncthreshold = 1
7 changing the directory paths and activating the
> CFLAG+=-DAHDI flag
>
> Next I executed "make install"
>
> Last, I copied the chan_ss7.so to /usr/lib/asterisk/modules/
use last svn
svn co http://svn.dicea.dk/chan_ss7/trunk
i'm not tested dahdi 2.2, i'm u
ich chan_ss7 (1.2,svn,1.1,..?)
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Actually at the moment Libss7 only supports ISUP layer, for SMS usage SS7 stack
should support SCCP layer too which as far as I've heard Mathew is
eager to implement in future.
http://www.dicea.dk/company/ss7_smsc
but this imho isn't for libss7
---
DTMFUP
it's fixed in svn http://svn.dicea.dk/chan_ss7/trunk
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Marek Cervenka
jabber - cerv...@njs.netlab.cz
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+multiple
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Marek Cervenka
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erisk 1.6.0.21.
switch to 1.6.1.x if you can
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asterisk-ss7 ma
ng smoothly.
===end text===
you can try:
- newer version of chan_ss7
- jitter buffer option in ss7.conf
[jitter]
jbenable = yes
jbmaxsize = 1000
jbresyncthreshold = 1000
jbim
sted in some ASR statistics like
http://www.ss7.pl/~jacke/patches/chan_ss7/chan_ss7-0.9.patch-3.gz
too
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Chan_ss7 seems not to be able
> able to setup link on mtp level. I got this permanently:
> ...
>
> Has someone already struggled with the same problem? Is there any patch
> for this?
i have the same problem. no solution
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s to "chop up" big packets in order to get the small RTP packets
going smoothly.
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et pending" is ok
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thout problems?
is there someone who can look at jb problem as paid contract? (netfors is
very busy now)
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as paid contract? (netfors is
> very busy now)
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ast
ed anything for me :-(
i have the same problem
asterisk 1.6.2.9+ chan_ss7 1.4.1
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;>> but it haven't changed anything for me :-(
>
> i have the same problem
> asterisk 1.6.2.9+ chan_ss7 1.4.1
the problem is probably somewhere in indication code. after downgrade to
1.6.2.1 ringback tone works
i'm trying comp
tone works
>
> i'm trying compare 1.6.2.9/1.6.2.1 indication code
patch is in
https://issues.asterisk.org/view.php?id=17372
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Marek Cervenka
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hi,
i'm posted example of loopback configuration for chan_ss7
it's useful for:
- stress testing
- performance testing
- debug
http://www.voip-info.org/wiki/view/Asterisk+ss7+loopback
-------
Mare
> Two weeks ago a working interconnection with another carrier went down,
> due to a reboot of our system. Since then chan_ss7 stopped working.
>
> When I load the module all looks fine:
send your configs
you can contact me on jabber if you want
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6.2.x
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he same.
>
change in l4isup.c
ALL "caller.id" to "connected.id"
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On 10/14/2011 02:30 PM, caio wrote:
>
>
> On Fri, Oct 14, 2011 at 8:45 AM, Marek Cervenka <mailto:cerv...@fpf.slu.cz>> wrote:
>
> On 10/12/2011 10:47 PM, caio wrote:
> > Hello,
> >
> > I have the following issue when calling from a s
hi,
there was demand for ss7 continuity check test for CLI
i have price from netfors.com for this function
it's too much for me. if there are interested other people please
contact me privately
bounty ends on 30.10.2011
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Marek Cer
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