Displaying 20 results from an estimated 6000 matches similar to: "IAX jitterbufer oddity"
2013 Jun 24
3
[PATCH v2] xen-netback: add a pseudo pps rate limit
VM traffic is already limited by a throughput limit, but there is no
control over the maximum packet per second (PPS).
In DDOS attack the major issue is rather PPS than throughput.
With provider offering more bandwidth to VMs, it becames easy to
coordinate a massive attack using VMs. Example: 100Mbits ~ 200kpps using
64B packets.
This patch provides a new option to limit VMs maximum packets per
2005 Oct 18
1
sip rfc bye violated?
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRel INVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6. Unhold SIP/2.0
7. Rx SIP/2.0 / 102 INVITE
8. CancelDestroy
9. Unhold SIP/2.0
10. Rx
2009 Sep 23
4
International Numbering plan ?
Hi
anyone know where i can find all internatinal numbering plan in csv and
for free or small price ?
thanks
Jpc
2009 Sep 23
3
Bringing people into a conference
G'day all, I'm using Asterisk 1.4 and am trying to work out a way to bring
people into a conference call. In the ideal scenario two people would be
talking and one of them would push some keys, then a phone number and then
the three of them would be in a conference. From there they should be able
to bring in other people as well.
This seems to be what the Asterisk n-way call HOWTO
2006 May 03
2
New jitter.c, bug in speex_jitter_get?
On 5/3/06, Jean-Marc Valin <Jean-Marc.Valin@usherbrooke.ca> wrote:
> > I must say I really like the generalized jitter buffer though :) It's a
> > cleaner and more flexible implementation and can more easily be adjusted
> > to contain additional information with each packet.
This looks interesting to tie into asterisk's jb and plc code as well.
2009 Aug 31
4
Inquiry:How to hide Caller Id
Dear All
Can you please do me favor and let me know how I can hide the subs number
being displayed on his phone when he goes off hook ? I mean when the subs
goes off hook he sees his assigned number on his phone and I need to disable
this feature . I don't know from which configuration file this feature is
coming so please let me know how can I disable it .
Regards
H.Motamedi
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2009 Oct 29
5
Dynamic DNS trunk
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP of the DNS, I have to reload and save the
configuration again so that asterisk recognize the new IP of the DNS.
Any idea how to automate such a thing? Or how can I keep asterisk to deal
with NAMES as NAMES, and IPs as IPs.
Let me know.
Thanks.
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2006 May 03
2
New jitter.c, bug in speex_jitter_get?
> Yes. Jean-Marc has made the API more similar.
>
> Jean-Marc: Have you looked at the API we have for the
> asterisk/iaxclient jitterbuffer?
Just did.
> It's pretty close to what you have now -- the major difference is that
> your jb still assumes it can "own" the data passed in -- it copies it,
> and it destroys it at will. With the API I put together,
2009 Oct 21
4
Concurrent calls including mysql taking lot of time for execution
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
optimized all possible parameters in my.cnf:
max_connection=1000
wait_timeout=60
query_cache_type=1
query_cache_limit=4M
query_cache_size=512M
interactive_timeout=120
2010 Sep 17
1
Attended Transfer does not release channels
Hi all,
i have the following setup
PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk
1.6.2.9 -> SIP -> agent
Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can initiate the transfer over a web interface
- it does generate a asterisk manager atxfer request...
So agent does initiate transfer - call
2009 Sep 10
2
ASR & ACD
Is there any program Asterisk users use to calculate ASR and ACD ??
Thanks for any comments.
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2016 Aug 04
2
Migration from samba3 to samba4 : PDC doesn't not appear in network
Rowland Penny a écrit :
> On Thu, 4 Aug 2016 12:12:42 +0200
> JB <jb at eikeo.com> wrote:
>
>> Hello,
>>
>> I'm trying to migrate an old PDC controller running samba
>> 3.0.4 to a more decent server. Now, I use samba 4.2.10 (from
>> debian/jessie).
>>
>> My smb.conf is :
>>
>> # Global parameters
>> [global]
>>
2009 Aug 08
1
30 Great free Asterisk applications
Hi, I was looking round on the Internet and saw there was no definitive
list of free applications available for use with Asterisk, so I thought
I'd compile a list for you all. If there's anything that you know of
that is actively maintained but not in the list below, let me know (bear
in mind I'm not including distros or Asterisk packagings in this list).
Hopefully there are a few
2015 Jan 29
1
JITTERBUFFER function
> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to
call
> > the JITTERBUFFER function?
>
> You only need to use the JITTERBUFFER function.
>
> The jbenable option will enable a jitter buffer on every channel
> created for that peer (or, if global, for every peer in the system).
> Depending on the version of Asterisk, it will also place the
2013 Feb 06
0
[PATCH 1/4] xen/netback: shutdown the ring if it contains garbage.
A buggy or malicious frontend should not be able to confuse netback.
If we spot anything which is not as it should be then shutdown the
device and don''t try to continue with the ring in a potentially
hostile state. Well behaved and non-hostile frontends will not be
penalised.
As well as making the existing checks for such errors fatal also add a
new check that ensures that there
2006 Oct 27
1
Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include "iax.voip1.conf"
#include "iax.renoir.conf"
The iax.voip1.conf file contains :
[VOIP1]
type=friend
2009 Aug 27
6
Measuring voice quality with Asterisk
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)?
Thanks
Klaus
2010 Jul 28
2
IAX authentication oddity - Known issue? Fixed?
Hi,
I had the following odd behaviour in Asterisk 1.2 - We are migrating
to 1.6, and I will re-test ASAP, though it is quite hard to replicate,
but I am curious to know whether it is a known IAX issue in 1.2.
We had 2 users in iax.conf:
[user1]
username=user1
secret=secret1
context=context1
host=iax.hostname.com
[user2]
username=user2
secret=
context=context2
host=dynamic
deny=0.0.0.0/0.0.0.0
2009 Dec 23
1
Rgraphviz on mac 10.6.2
Rgraphviz Install works fine (http://www.bioconductor.org/packages/release/bioc/html/Rgraphviz.html)
Latest version of graphviz is installed as well
however I get following error when loading Rgraphviz (on Mac 10.6.2)
Error in dyn.load(file, DLLpath = DLLpath, ...) : unable to load shared library '/Users/jb/Library/R/2.9/library/Rgraphviz/libs/i386/Rgraphviz.so':
2009 Aug 12
3
Asterisk + CDRTool
Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe can suggest another CDR GUI ?
regards.
Harry
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