On Tue, 23 Mar 2010, ayodele abejide wrote:
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Mon, 22 Mar 2010 17:54:27 -0600
Subject: Re: [asterisk-users] Can I call myself on the same machine
On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote:
I am a newbie to
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,
On Mon, Mar 22, 2010 at 09:38:26PM -0400, sean darcy wrote:
1.6.2:
-- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1,
1...@default,u) in new stack
-- DAHDI/4-1 Playing
'/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en')
[Mar 22 17:15:46]
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.
21 mar 2010 kl. 18.22 skrev Philipp von Klitzing:
Hi Olle!
The work I started during Christmas - Named ACL's - is a starting point
that other developers can use to develop all kind of schemes.
http://www.voip-forum.com/asterisk/2010-01/manageable-access-control-lists
-asterisk-nacls/
22 mar 2010 kl. 14.54 skrev Kevin Sandy:
On 3/21/2010 4:05 AM, Olle E. Johansson wrote:
17 mar 2010 kl. 16.37 skrev Kevin Sandy:
We're having an odd issue with codec negotiation from one of our
SIP providers. Here's the basic situation.
We receive an invite from them advertising
[020202020202]
context=mgcp
host=dynamic
canreinvite=no
dtmfmode=rfc2833
nat=yes
threewaycalling=yes
transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to
transfer callwaiting=yes ; this might be a cause of trouble for ip10s
cancallforward=yes
line=aaln/1
Order of variables is
Hi,
We are trying compile dahdi on amazon vertual instance.
When we are compiling dahdi we receieve following error.
You do not appear to have the sources for the 2.6.21.7-2.fc8xen kernel
installed.
We are helpless on getting this 2.6.21.7-2 sources. Please help to get this
compile.
--
On Tue, Mar 23, 2010 at 06:27:40PM +0530, Vidura Senadeera wrote:
Hi,
We are trying compile dahdi on amazon vertual instance.
When we are compiling dahdi we receieve following error.
You do not appear to have the sources for the 2.6.21.7-2.fc8xen kernel
installed.
We are helpless on
Hi there , are you using any king of Iax trunk or Duguim interface on this VM?
Because if is just for sip you dont need dahdi you can compile asterisk and
work on it.
Daniel Abreu
On 23 Mar 2010, at 12:57 PM, Vidura Senadeera wrote:
helpless on getting this 2.6.21.7-2 sources. Please help
- Daniel Leite de Abreu dlab...@gmail.com escreveu:
Hi there , are you using any king of Iax trunk or Duguim interface on
this VM?
Because if is just for sip you dont need dahdi you can compile
asterisk and work on it.
He will need DAHDI if he plans on using MeetMe().
Also, internal
While working with Rhino hardware I was told by their technical support that
no virtual machine software gives access to the PCI bus, so using zaptel or
dahdi is not an option over the virtual machines. Although somebody has said
otherwise on this list but make sure you actually have access to the
On Mon, Mar 22, 2010 at 7:56 PM, Rafael Prado Rocchi
pr...@practis.com.brwrote:
Hi, it's not that simple.
It requires deep modification on asterisk and dahdi sources to work the way
you want.
Why? I must confess I still don't quite understand what he wants, from what
I've read the legacy pbx
For that ztdummy does the work fine without worrying about enabling the
actual hardware.
On 2010-03-23 9:49 AM, VinÃcius Fontes vinic...@canall.com.br wrote:
- Daniel Leite de Abreu dlab...@gmail.com escreveu:
Hi there , are you using any king of Iax trunk or Duguim interface on
this VM?
My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function
at:
return (*vp-u.s == 0 || (to_integer (vp) vp-u.i == 0));
My gdb trace is at:
http://pastebin.com/raw.php?i=hmhzZxye
Other examples here:
http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html
Thanks Zeeshan,
In fact,i have RealTime configured and working...
What i want is to make an upgrade of libpri and wanpipe at least, asterisk
and zaptel will be like i have now...
Do you think that recompile/upgrade this softwares version will produce a
problem? what steps should i do?
Is it
Dear All
i'm planning to develop for a customer a particular implementation of Asterisk.
The aim of the project is to share different users between different
Asterisk inbound call center .
I'm planning to have a sync for some of the QueueMemberStatus
informations between all the nodes, then a
Hi,
Do clone the existing server in any case. I am not sure about libpri but
zaptel I have recompiled various times without recompiling asterisk, so I
think it should be ok with libpri too. Wanpipe doesn't interact with
asterisk, but with zaptel only so it should not be a problem to update it.
Hello list,
what can I do to minimalize the jitter in SIP-calls at server level ?
If at local network level, there is a VoIP-router and their is a
physical network dedicated to IP-phones, but there is still jitter.
When using a Hosted Asterisk server, which settings on the
Asterisk-server can
Hi Everyone,
I have tried to set the box to DMZ and also tried to port forward 5060
TCP/UDP and 1/2 UDP to the server IP but it's no use and there is a
no audio issue. I am pretty certain it's a NAT issue as the sip call
establishes. I also made a succesful IAX2 call through IAX trunking
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com wrote:
My Asterisk 1.2.40 process crashes
regularly in the is_zero_or_null function at:
return (*vp-u.s == 0 || (to_integer (vp)
vp-u.i == 0));
My gdb trace is at:
http://pastebin.com/raw.php?i=hmhzZxye
Other examples here:
Friends,
Daniel and I are running a Kamailio SIP Masterclass this week in Berlin. When
travelling around like this, we often invite the community to come and meet us
in a nice restaurant. We offer good company and fun discussions about Kamailio,
SIP-router.org and Asterisk - but the drinks and
Running * version 1.6.1.17.
My meetme conferences automagically disconnect users approximately 5-15
seconds after the user is connected. This occurs regardless of whether
music on hold is active or not.
[Mar 23 11:34:36] -- Executing Macro(SIP/SDN_TMCKEE-00e9,
confroom,1808)
[Mar 23
Hi ,
I had some problems in the past with sip trunks, asterisk-users Digest, Vol
68, Issue 4, message 6, and had a reply (message 9) saying that It could be
a dns issue.
Well today I had a problem again with sip module and it really seams a dns
issue.
I have an asterisk, version 1.4.26.1,
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
Thank you !!!
Alejandro
--
On 24/03/10 8:41 AM, Alejandro Cabrera Obed wrote:
Dear all, I have an Asterisk SIP server in a LAN environment and I want
your opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN
voip calls ???
It basically comes down
Doesn't really matter unless you want to use zaptel lines on one end and
VoIP lines on the other end, and want to avoid mismatch between your telco
and your IP phones codecs. Transcoding between aLaw and uLaw in my
experience can degrade voice quality. If you are using zaptel, and your
telco
2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com:
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
That depends most on
fwiw, they both 'sound' the same, so if your question was specifically about
not pest-terminated calls it still doesn't matter. :-)
For that you could use something like g.722 provided your endpoints support
it. In that case it really DOES make a big difference and the bandwidth is
the same.
On Tuesday 23 March 2010 11:53:03 Vieri wrote:
--- On Tue, 3/23/10, Vieri rentor...@yahoo.com wrote:
My Asterisk 1.2.40 process crashes
regularly in the is_zero_or_null function at:
return (*vp-u.s == 0 || (to_integer (vp)
vp-u.i == 0));
My gdb trace is at:
On Tue, Mar 23, 2010 at 1:41 PM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
Does anyone know the rationale behind why deny/permit values can not be
specified in 'general' setting of sip.conf iax.conf
In other words, if I want to deny everyone, then allow selectively permit
specific hosts or subnets, I can't do so without first deny'ing all in EVERY
user/peer
Karl Fife wrote:
Naturally I can accomplish this using templates, but it DOES seem a bit odd
since [general] is in essence ALREADY a 'template' of sorts for all
parameters not otherwise specified.
You should use templates; the [general] section never should been an
implicit template, but it
Karl Fife wrote:
Naturally I can accomplish this using templates, but it DOES seem a bit
odd since [general] is in essence ALREADY a 'template' of sorts for all
parameters not otherwise specified.
On Tue, 23 Mar 2010, Kevin P. Fleming wrote:
You should use templates; the [general]
Steve Edwards wrote:
It may not be as intended, but from a user standpoint, it seems logical
and convenient to establish policy in [general] and make exceptions in
the entities as needed.
Right... for when you have one policy. When you have two policies, each
that apply to a dozen or more
I see.
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 23, 2010 4:06 PM
Subject: Re: [asterisk-users] permit/deny in sip.conf iax.conf
Karl Fife wrote:
Steve Edwards wrote:
It may not be as intended, but from a user standpoint, it seems
logical and convenient to establish policy in [general] and make
exceptions in the entities as needed.
On Tue, 23 Mar 2010, Kevin P. Fleming wrote:
Right... for when you have one policy. When you have
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know is if safe_asterisk has something to be with this?
This is what i have on my server:
[r...@mypbx ~]# ps -A | grep
On Tue, 23 Mar 2010, Danny Dias wrote:
This safe_asterisk could be the cause of my problems? how does it works?
how can i activate it?
safe_asterisk is a script that usually lives at /usr/sbin/safe_asterisk.
The script runs in the background. If it detects that Asterisk died, it
can send
Hi All,
I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit
a snag with the Grandstream HT502. It only seems to nail up a session at
9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm
using the same equipment in the same configuration, just
I am running Asterisk and using Answer machine detection with call files on
a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD
is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over
50,000 outbound calls last week, and 70% said NOTSURE).
I have a
hi, all
i use Queue() to call a Mobile phone, there is only one mobile phone
in the queue. even if the mobile phone shut down, Queue() is ring in
the cli verbose
as mobile phone is normally working. what i want to see is if the
mobile phone is shut down.
queue() will end immediately to tell on
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls
are not being picked up. I don't find anything unusual in asterisk
log. I am clueless where I should look. I also find
zapata-additional.conf empty. The trouble started when the system was
accidentally shut down and rebooted.
Try the same as in
http://lists.digium.com/pipermail/asterisk-users/2010-March/246316.html
just make sure to add this in the [channels] context ;)
Hope it helps.
Alyed
2010/3/23 Zhang Shukun bit...@gmail.com
hi, all
i use Queue() to call a Mobile phone, there is only one mobile phone
in
Hello,
Please Confirm if the dahdi/Zaptel service is running .
check your channels status.
On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman brama...@gmail.com wrote:
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls
are not being picked up. I don't find anything unusual in
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