Displaying 20 results from an estimated 20 matches for "chan_ss7".
2009 Mar 20
1
chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===================================================
sta...
2007 Nov 21
0
chan_ss7 0.10.1
hi,
i'm added another patch to chan_ss7
it's from Denis Smirnov http://download.seiros.ru/SeirosPBX/chan_ss7/
New in version 0.10.1 (community version)
- support for more than 256 channels
- zap style addressing
http://download.seiros.ru/SeirosPBX/chan_ss7/
http://www.freevoice.cz/chan_ss7/chan_ss7-0.10.1.tar.gz
md5sum a3ca3031f...
2012 Jul 12
0
chan_ss7 quick patch to enable RBT
Hello everyone,
I am trying to apply
this<http://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diff>patch
on chan_ss7-2.1.0 for RingBack tone but its not accepting and
throwing errors:
Hunk #1 FAILED at 704.
Hunk #2 FAILED at 715.
I have done the patch modifications manually in l4isup.c
There is just one question, how do I pass the RB file-to-play on an SS7
channel via asterisk?
--
Thanks.
P.S. here is the sou...
2010 Jan 21
0
chan_ss7 or libss7, which is more stable?
Hi, I?m trying to use SS/ in Asterisk.
I'm thinking in chan_ss7 and libss7, and I want to know some other
experience with this.
Thanks!
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2010 Mar 23
0
[asterisk-ss7]Chan_ss7 issue
Dear all,
Do you have come acrross with this issue. My ss7 link get fluctuating. It
use chan_ss7 version 1.0.95-beta.
I have 8 E1s running on a DL380 server with Digium E1 cards ( 4 port cards).
This enable to have calls from sip to ss7 and vice versa. However ss7 links
are not stable.
linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4,
sentseq/lastack: 127/127, total 40...
2011 Dec 28
0
Chan_ss7 clustering config with single point
Hi team,
Can any one share with me clustering configuration file SS7.conf for single pointcode with four slc. two different machine each host having 2 slc respectively.
Thanks
Vinod Dharashive
Sent from BlackBerry? on Airtel
2006 Sep 14
3
One way audio problem on gateway to PSTN after some time, no NAT involved
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN
What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is...
2010 Mar 23
1
chan_ss7 issue
Dear all,
Do you have come acrross with this issue. My ss7 link get fluctuating. It
use chan_ss7 version 1.0.95-beta.
I have 8 E1s running on a DL380 server. This enable to have calls from sip
to ss7 and vice versa. However ss7 links are not stable.
linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4,
sentseq/lastack: 127/127, total 4034145216, 4031118560
linkset siuc, li...
2007 Dec 02
1
setting up two asterisk server as ss7 back to back.
I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all
went okay. 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 signaling link does not align.
i have my configs for ho...
2006 Mar 31
1
transcoding g723 or g729 on asterisk
...bother the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice versa.
I will be glad if someone can throw more light on this for me.
Goksie
-----Original Message-----
From: asterisk-ss7-bounces@lists.digium.com
[mailto:asterisk-ss7-bounces@lists.digium.com] On Behalf Of Kai Militzer
Sent: Monday, March 27, 2006 3:19 PM
To: aste...
2006 Mar 31
0
Transcoding on asterisk
...her the list too much. However, after reading I discover
I don?t have a clear cut way of doing transcoding.
Can somebody direct me to where I can get document to get this transcoding
done.
My set up
>From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same
asterisk) g711 to chan_ss7] -----> [pstn]
And vice versa.
I will be glad if someone can throw more light on this for me.
Goksie
-----Original Message-----
From: asterisk-ss7-bounces@lists.digium.com
[mailto:asterisk-ss7-bounces@lists.digium.com] On Behalf Of Kai Militzer
Sent: Monday, March 27, 2006 3:19 PM
To:...
2006 Apr 06
0
What Media Gateway (connected via SS7) do you use
...asterisk-biz@lists.digium.com
Subject: [asterisk-biz] What Media Gateway (connected
via SS7) do you use
Hello everyone,
I am doing some research for my company (which does
not want to be named therefore the gmail. I am sorry
for that) which carrier grade media-Gateways
(read: _not_ asterisk with chan_ss7 or ss7box, etc. *)
are used by others to connect their asterisk (or any
other SIP-Proxy,
- Router, -PBX, etc.) to the PSTN via SS7.
We are currently looking for something that can be
connected via SS7 in Europe, scales good (i.e.
starting with 4-E1s, but with no limits to add
more) and supports
*...
2010 Jun 11
1
WARNING message when play
When I use an eagi script when play a message appear a lot of warning
messages, but it play very well
I?m using
Asterisk 1.4.32
dahdi-linux-2.3.0.1
chan_ss7-1.4.1
Any ideas??
-- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0)
[Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write()
failed: Broken pipe
[Jun 11 18:12:45]...
2007 May 02
6
allowing call every 15mins
...reduce.
I want all call to get to voice prompt except that that enter when
minutes is 15, 30, 45, 60(in multiples of 15 minutes).
how can i achieve this and what application can i use to get this done.
I will be glad, if someone can give me a hint on this.
i have asterisk-1.12.1
zaptel-1.9.1
chan_ss7-0.8.4
Goksie
2008 Sep 29
0
AGI defunct processes + GSM Playback - HELP!
Hello.
I've just installed
asterisk-1.4.21.2
zaptel-1.4.12.1
chan_ss7-1.0.10
libpri-1.4.7
I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers.
My OS: Ubuntu 8.04 Server
Kernel: 2.6.24-16-server
I am getting a choppy GSM playback and too many defunct AGI processes when
channel closes.
i am using Perl or PHP, also 'agi-test.agi' going to defunct too......
2008 Nov 01
0
asterisk 1.2 and Dial with LIMIT_WARNING_FILE
...xten => _09049.,113,Dial(${TYPE}${DESTINO}|30|L(30000:10000))
works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the
end of the call.
doesn't work on my Asterisk 1.2.13, it hungs 10 seconds before the end
of the call, just when it has to beep
both of them have the same chan_ss7 and the beep.gsm in the correct place.
Do you have any idea of what is happening?
Thanks in advance..
Rafael Visser
2012 Sep 12
3
kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
I have a server with an asterisk ss7 link connected to a Siemens working
well for over a year.
A few days ago I started having problems with signaling.
I found the following logs in / var / log / messages
Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to
TE2/0/2
Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to
TE2/0/1
Sep 12 11:49:26 call3 kernel:
2006 May 03
1
my asterisk crashed
the gdb of the core taken from the asterisk as the time of crash is as below
I run asterisk-1.2.5 on fedora core 3 with chan_ss7
can someone help out?
#0 ast_var_name (var=0x1) at chanvars.c:71
71 if (var->name[0] == '_') {
(gdb) bt
#0 ast_var_name (var=0x1) at chanvars.c:71
#1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46
"OUTBOUND_GROUP") at pbx.c:5904
#2 0xf5bbe...
2008 Jan 23
3
asterisk optimalization
hi,
i'm testing asterisk 1.4/1.2 in the following scenario
centos5/cpu quad xeon E5335 2.0Ghz
- test clients behind nat
- 1500+ testing instances - reregister option from 1min to 1hour
- qualify set to 5000
top shows over 100% cpu. cpu cores sometimes go to 95%
with htop i see ~16threads but only one child have ~95% cpu
(how i can get info about that thread? what he is doing?)
what is
2010 Sep 17
3
Sangoma A108 PCIe V2.0
Hi
Does Sangoma 8-port card A108 support PCIe version 2.0 ?
The card is here
http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a108.html
And we want to use 3 such cards in this motherboard because it has 3 PCIe
slots of version 2.0
http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.htm
Is this a good idea ? Do you have any experience