, "peterpet" wrote:
> Why i need to change telco's point codes configuration?
> My PC is 5450 ?!
>
>
>
> On 12/17/2013 06:34 PM, Vashkar Chatterjee wrote:
>
> try to put the value like this
>
> pointcode = 1890
> adjpointcode = 1802
> defaultdp
try to put the value like this
pointcode = 1890
adjpointcode = 1802
defaultdpc = 1802
On Tue, Dec 17, 2013 at 10:24 PM, peterpet wrote:
> Hi everyone,
>
> i have a really strange problem. Signaling is looking OK, but not.
> INSERVICE but DOWN. My Telco did not receive SLTM messages from me
Your end point code should be in decimal of 0x762 hex
On Dec 17, 2013 11:24 AM, "peterpet" wrote:
> Hi everyone,
>
> i have a really strange problem. Signaling is looking OK, but not.
> INSERVICE but DOWN. My Telco did not receive SLTM messages from me.
>
> *ss7 show linkset 2*
> SS7 flags: 0x0
Please explain your requirement and be more specific.
On Mon, May 13, 2013 at 5:15 PM, Nyamul Hassan wrote:
> No one knows if this is possible?
>
> Regards
> HASSAN
>
>
> On Thu, May 9, 2013 at 4:54 PM, Nyamul Hassan wrote:
>
>> Hi,
>>
>> I was reading upon SS7 call forwarding, and there does
There sbould be no relation in voice and sig as ss7 is a ccs type
signalling. So voice circuits can go to any node.
Vashkar.
On Apr 22, 2012 6:19 PM, "her Garcia" wrote:
> Hi, thanks for your replies. I ´ll try your suggestions. The reason I have
> differents adjpointcode and de
Try putting sane point code in the adpointcode as defaultdpc
On Apr 20, 2012 7:16 PM, "her Garcia" wrote:
> Hi, everyone. I am working on asterisk+ss7.
> When I try to make a call, the call connects but I have no audio or see no
> progress in the debug.
>
>
> -- Executing [111536972876@incomi
CIC mappings is not correct. Thats why you are not getting any audio.
-Vashkar
On Mar 17, 2012 1:34 AM, "Marcus Vinicius" wrote:
> Hi,
>
> I have an linkset like this:
>
> linkset=1
> pointcode=1
> defaultdpc=2
> adjpointcode=2
>
> context=from-pstn-01
Try to debug ss7 and check after which ss7 message it happens.
It happens sometimes with bad physical link or even when message is not
understood.
-Vashkar
On Mar 13, 2012 12:46 AM, "Marcus Vinicius" wrote:
> Hi,
>
> I'm testing libss7 and I'm getting some
Check transmission for faults. Do you get BAD FCS HDLC abort? Check for
the line error, code violation etc from dahdi maint tool
-Vashkar
On Feb 15, 2012 1:04 PM, "Bharat Lalcheta" wrote:
> Thanks for reply.
> Telco enabled crc and first server working fine on it.
>
> O
Checm if telco had crc enabled or not. If not turn off crc
On Feb 15, 2012 12:45 PM, "Bharat Lalcheta"
wrote:
> Hii,
>
> I have 2 servers with dual core 3 GHz dual cpu with 2 GB RAM and having 2
> * TE410P digium, total 8 port for inbound calling.
>
> using asterisk-1.6.2.9, dahdi-2.5, libss7 SVN
Try signal transfer point.
On Feb 11, 2012 1:18 PM, "Mehdi Shirazi" wrote:
>
> Hi
> I want to put one signaling probe between two TDM Exchanges(local =>
> transit) and just pass signaling link through this probe (voice channels go
> directly with 150E1) , this probe should transparently send all
hi,
check if that the caller.id has been changed to connected.id in the
l4isup.c file first. because it will not reflect the correct caller id at
the first place... then look forward to the issue
-Vashkar
On Sat, Jan 21, 2012 at 11:54 PM, Marek Cervenka wrote:
> hi,
>
> i fou
Post your extension.conf
Vashkar
On Oct 13, 2011 2:49 AM, "caio" wrote:
> Hello,
>
> I have the following issue when calling from a sip endpoint to a pstn
> number.
>
> i don't know why the chan_ss7 is taking same values for called and calling
> party numb
Check your physical connections
On Sep 28, 2011 2:01 AM, "Ryan Crowder" wrote:
> This is what I keep getting. Can someone help me debug? I have a single
> T1 running ss7 to this same switch in another box but I haven’t been able to
> get this up.
>
> Thank you!
>
> ** **
>
> [Sep 27
It works finr
Can anyone tell is there any way to capture the MTP3 and ISUP messages on
libss7+dahdi?
-Vaskar
On Thu, Sep 15, 2011 at 6:22 PM, Torrey Searle wrote:
> As there seems to be some confusion on how to get dahdi_pcap up and
> running, I've decided to repost the current install proced
Check the jumper settings on the e1 card that sets e1 or t1
-Vashkar
On Jul 24, 2011 9:25 PM, "Dave George" wrote:
> Seems the card on span 9 switched from T1 to E1 after power failed. How do
> I set it back to T1 using config?
>
>
>
> Dave
>
>
>
>
try putting prematureaudio=no
the flag "yes" acctually is a reverse. means that if you put "NO" then the
system passes the pre-mature audio
The progressinbad should be "never"
-Vashkar
On Wed, Jun 22, 2011 at 3:51 PM, Nyamul Hassan wrote:
> Hi,
>
> We
.
-vashkar
On Sat, Jul 31, 2010 at 1:10 AM, Nyamul Hassan wrote:
> Yeah, it can be a bit daunting.
>
> The choice between libss7 or chan_ss7, is still a matter of debate. We had
> pretty good luck with chan_ss7. Using a custom made server with quad core
> Xeon processor and 4 GB of RAM
stable with libss7 current version and chan_ss7
1.3 or higher. Place all the cards on diffrent cpu to handle interrupt.
-vashkar
On Sat, Jul 31, 2010 at 12:53 AM, Nitesh Divecha
wrote:
> Thanks Hassan,
>
> Yea, I have been reading the Asterisk-SS7 threads but everyone has their
>
can look for SIGTRAN. I have no idea if asterisk can do that or not
-vashkar
On Mon, Jul 5, 2010 at 5:04 AM, mosbah abdelkader
wrote:
> Thanks for your great ideas.
>
>
> Can someone describe a use case of ISUP over IP configuration. Is it
> possible for example to us
Try removing the hw echo card. Normally axe 10 should handle the echo issue.
-Vashkar
Sent from my mobile device.
-Original Message-
From: Jean Cérien
Sent: 31/03/2010 12:55:35 am
Subject: [asterisk-ss7] echo on a SS7/E1
Hello
I have the following setup:
Ast1.6.1.18+Chan-SS7 1.3
are you having two C7 configured ? I mean on ts 16 and ts 47 ?
On Fri, Dec 18, 2009 at 10:52 AM, bipin singh wrote:
> hi
>your sustem.conf is wrong
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> try this
>
>
> On Wed, Dec 16, 2009 at 11
its true that the cluster is not required. the same box will work accross
the network as the signalling point.
cluster is for a backup N7 link
Be sure to add host entries on the system.
-Vashkar
On Fri, Jul 17, 2009 at 10:58 PM, Nyamul Hassan wrote:
> I understand that it should work.
it must work on 8 or more E1's
only thing is to get the signalling server from all the system and using
cluster
-Vashkar
On Fri, Jul 17, 2009 at 3:28 AM, Nyamul Hassan wrote:
> Hi Antoine,
>
> Did you get around to using this setup with full 8 E1s?
>
> It would be very
gcc -MM -E -I/usr/include/zaptel -I/usr/include/asterisk
this should point your make file
* -I/your zaptel source/kernel -I/your asterisk source/include*
make sure to edit this line. then do make clean and make
should build your chan_ss7.so mtp3d.so and others.
-Vashkar
On Wed, Jun 24, 2009
chan_ss7 can be compiled on 64 bit
On Tue, Jun 23, 2009 at 1:59 AM, caio wrote:
> Hi,
> I'm trying to compile chan_ss7 sources under Gentoo amd64 platform but it
> fails.
> Here I attach you the "make" output. I don't know if chan_ss7 is 32-only
> compliant or if I missing something causing this
hey bro you did a grat job. at least your trick works :-).
Problemis to find out ss7 spec on the internet is very troublesome. Most of
them you may need to buy.
Your addon can be used with the people who are working with ANSI variant of
SS7 :-)
-Vashkar
On Mon, Jun 8, 2009 at 8:11 PM, Ruddy
if you are using chan_ss7, put the keyword
variant and its value to ANSI
look for the same thing if you are using DAHDI and libss7
it should work.
-vashkar
On Thu, Jun 4, 2009 at 4:50 AM, Ruddy Gbaguidi wrote:
> I don’t think so.
>
> Do you now where I can find specs of SS7 ANS
Can you ask your operator to use ITU-WHITE instead of ANSI? I think that
will solve the problem.
On Tue, Jun 2, 2009 at 12:20 AM, Ruddy Gbaguidi wrote:
> Hi all
>
> We had a SS7 link with our provider here in Canada. Everything went well.
>
> But they are saying that asterisk is missing the SLC
normally when you compile the chan_ss7, it will compile the mtp3d. You do
not need to compile it separately.
-Vashkar
On Tue, Jun 2, 2009 at 5:45 PM, Lakshmi Narasimhan .R
wrote:
> > Make sure that that you have given the right path to asterisk and
> > zaptel lib in the make file.
Make sure that that you have given the right path to asterisk and
zaptel lib in the make file.
On 5/30/09, Lakshmi Narasimhan .R wrote:
> Hi,
>
> I am trying to compile chan_ss7 1.0.0 under asterisk 1.2 and zaptel
> 1.2 with mtp3d support. I am getting the below error.
>
> [r...@hostname chan_ss7
Hi Sumon.
we can provide you the solution. If you are interrested. you can contact me.
Vashkar
On Sun, May 24, 2009 at 3:45 AM, Sumon Ahmed wrote:
> Dear All,
>
> One of my Friend setup an Asterisk Box as SS7 Termination Gateway.From my
> netflow report, I can see only UDP pac
the reason.
Also check if your operator accesspets the leading '880' to be dialed or its
without them. May be '01720039748' you can try.
If they want to strip digits, set the extentions.conf to
exten => _X.,3,Dial(DAHDI/g1/${EXTEN}:2)
.
-Vashkar
On Thu, May 14, 2009 a
not? if it is possible please
light me the path.
Regards,
Vashkar
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