Ask for the trace from you operator, so you will find out
what number and other properties you send out.
On Thursday 18 March 2010, Jorge Antillon wrote:
> Hi,
>
> I've been trying to setup SS7 completely down
> here in Costa Rica for a week now and I am thinking I've
> run out of reso
On Wednesday 24 Mar, 2010, at 6:19 PM, marek cervenka wrote:
> i'm updated FAQ on http://www.voip-info.org/wiki/view/Asterisk+ss7+faq
> from some old source
>
> try activate jitter buffer
> http://www.voip-info.org/wiki/view/Asterisk+ss7+configuration
> [jitter]
> jbenable = yes
> jbmaxsize = 1000
> I am using asterisk-1.6 with chan_ss7-1.3 on sangoma cards (a104de.)
> Everything is working fine but I notice many time the following
> message on the logs:
>
> NOTICE[22852] l4isup.c: Write buffer full on CIC=12 (wrote only 160
> of 240), audio lost (suppress 1).
> NOTICE[21785] l4i
Thank you for everithing Attila,
you are the best :)
Attila Domjan wrote:
> On Wed, 2010-03-24 at 15:18 +0200, peterpet wrote:
>
>> Hi Attila,
>> now everithing is ok, but why now this is not working with not included
>> timers in configuration?
>>
> In my version:
>
> 1. startin
On Wed, 2010-03-24 at 15:18 +0200, peterpet wrote:
> Hi Attila,
> now everithing is ok, but why now this is not working with not included
> timers in configuration?
In my version:
1. starting dialplan on timeout works via isup_timer.digittimeout timer
expiry
2. for example, you send REL, the
Hi Attila,
now everithing is ok, but why now this is not working with not included
timers in configuration??
I have 5 linksets , and in chan_dahdi linksets is without timers and
everithing is ok, but with this
i need to setup timers :( why?
and _X. -> this is just a example, in real a hav
Hi,
you missed set up the timers
when you are using _X. like patterns, and you don't get end of pulse, on
isup_timer.digittimeout expiry will start the execution the dialplan.
set up all the timers, and avoid using _X. like patterns. When you know
the exact digitlengths use:
_0
and
hi,
this is trace from call :
Network Indicator: 2 Priority: 0 User Part: ISUP (5)
[ 85 ]
OPC 118 DPC 5176 SLS 11
[ 38 94 1d b0 ]
CIC: 11
ss7-mtel-fix*CLI[ 0b 00 ]
Message Type: IAM
[ 01 ]
--FIXED LENGTH PARMS[4]--
Nature of Connection Indic
Hello,
I am using asterisk-1.6 with chan_ss7-1.3 on sangoma cards (a104de.)
Everything is working fine but I notice many time the following
message on the logs:
NOTICE[22852] l4isup.c: Write buffer full on CIC=12 (wrote only 160
of 240), audio lost (suppress 1).
NOTICE[21785
Thanks!
I had the same issue in South Africa, but eventually got the network on the
remote end to set up different CIC ranges to work around the issue in
chan_ss7. It's good to have a fix for chan_ss7 that resolves this.
--Greg
- Original Message -
From: "Robert Verspuy"
To:
Sent: Wed
contact a...@netfors.com
On Wed, Mar 24, 2010 at 10:18 AM, Robert Verspuy wrote:
> All,
>
> Here in the Netherlands we have a SS7 connection with the biggest local
> telco.
> This local telco has 2 redundant networks, and uses different point
> codes for signaling and the audio streams.
>
> we
All,
Here in the Netherlands we have a SS7 connection with the biggest local
telco.
This local telco has 2 redundant networks, and uses different point
codes for signaling and the audio streams.
we have 6 E1's connected on one server.
port 1 -> network A, with signaling link
port 2 -> network B
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