Check the sip.conf.sample. 1.8 has several options related to the SIP UPDATE
support.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan
Sent: Monday, March 12, 2012 10:10 PM
To: asterisk-users@lists.digi
Hi guys,
I am working on a setup where I have an Asterisk (1.6.2.7) with a SIP trunk
to a provider's softswitch(IMS). Trunking works all ok, calling out and
calling in. Except for the remote softswitch(IMS) will send UPDATE
command(according to RFC 3311) and Asterisk does not accept and thus the
ca
On 03/12/2012 06:52 PM, Markus wrote:
Now, if the "49" route of the first provider is cheaper, my
system (a2billing) will still use the more expensive "4930" code
because it is more specific.
There is a great deal of wisdom in this approach that you may wish to
consider carefully before aband
Hi,
this question is not Asterisk specific, but since there are so many
experts present on this list, maybe its OK to ask anyways.
I'm having a hard time "normalizing" rate sheets from different
providers. What I mean with this: the goal is to always get the cheapest
rate for a given destina
On 03/12/2012 03:38 PM, Steve Edwards wrote:
On Mon, 12 Mar 2012, Amit Patkar wrote:
What will be impact on no of session when G729a is used?
Assuming that transcoding is involved; if all the system is doing is
passing through G.729A media streams, and recording them in unmixed
G.729A form
On Mon, 12 Mar 2012, Amit Patkar wrote:
I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many
concurrent SIP sessions I can run from single instance of Asterisk on
this server? I wish to use G711 codec with echo cancel. And all calls
needs to be recorded.
What kind of capac
Che
Switch to a Grandstream HT-701 or HT-502 and see if you get better results.
The HT-286 has been discontinued for some time and the firmware has some
major T.38 issues.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: "Chet W.
I have been testing Fax For Asterisk with an analog fax sitting behind a
Grandstream Handytone 286. I have been sending a number of faxes but I find
that they send extremely slow. In my most basic test of sending from the HT286
straight to a tiff image
on Asterisk on the same subnet I am seeing
There are no such statistics. Your usage patterns are unique to you and depend
on many factors. If you must look for the information then look in the mailing
list archives or on voip-info.org.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Hi
Can someome give tested and proven information on Asterisk capabilities?
I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many
concurrent SIP sessions I can run from single instance of Asterisk on this
server? I wish to use G711 codec with echo cancel. And all calls need
On 03/12/2012 11:10 AM, Larry Moore wrote:
Check Start & End RTP Ports match values set in /etc/asterisk/udptl.conf
for udptlstart & udptlend
This is unnecessary; the two endpoints are free to use different port
ranges if they wish, it won't make any difference.
--
Kevin P. Fleming
Digium,
On 12/03/2012 10:53 PM, Ishfaq Malik wrote:
Thanks for the input so far. I'm going to keep plugging away and if
anyone has any insights, they will be gladly appreciated. Ish
In SIP Account Configuration on Draytek;
Set Voice Active Detect to Off
In Phone Settings on the Draytek;
Enable Symm
On Mon, 2012-03-12 at 14:53 +, Ishfaq Malik wrote:
> On Mon, 2012-03-12 at 09:11 -0400, Bryant Zimmerman wrote:
> > Looking at the information you have sent in this posting in certainly
> > appears that the 'f' option has indeed helped however you have
> > another
> > matter to overcome.
> >
To the best of my knowledge, your best options, not necessarily in order
are:
1. Vestec ASR
2. Lumenvox ASR
3. google ASR (there was a good post in February about how to use this)
4. Sphynx ASR
Options 1 and 2 are/were "recommended" by Digium.
-Original Message-
From: asterisk-users-boun.
Hi,
I am looking (for the best) solution to recognize *german* words or
simple phrases with a given number of words (eins, zwei drei etc. or
hauptmenü, zurück etc.). Can somebody give me a good link? Can I find
external service providers who can be accessed via ASR()?
Best regards,
-Thorsten
.
Now getting the following
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [200@local:1] Goto("SIP/588-000b", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-000b", "")
-- Executing
for using this application it is enough to put this in
chan_dahdi.conf ?
...
context=incoming
facilityenable=yes
transfer=yes
--- On Sat, 3/10/12, Karsten Wemheuer wrote:
From: Karsten Wemheuer
Subject: Re: [asterisk-users] DAHDISendCallreroutingFacility
To: "Asterisk Users Mailing List - No
Asterisk's peer matching changes between releases. Your best bet is to try it
and see.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asgaroth
Sent: Monday, March 12, 2012 10:22 AM
To: asterisk-users@lists.d
Hi,
On 12/03/12 13:48, Eric Wieling wrote:
Have you tried permit/deny on the peer?
No, I've not tried this, however, will those entries be checked if the
inbound call is not matched against the peer that those settings matched
with?
I'll have to research this option a little more.
--
___
Greetings back :)
Greetings,
That will depend on my SIP providers, I'm not sure if they swap their
IP's indeed,and send calls down my way with alternating IP's,
perhaps they're "smart" enough to only
send calls down my way with the same IP that was bound with the
registration request to begin w
Have you tried permit/deny on the peer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Guy Gold
Sent: Monday, March 12, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multi-reco
On Mon,Mar 12 01:33:PM, Asgaroth wrote:
> Are you saying that the above scenario is working for you for
> incoming calls?
Greetings,
That will depend on my SIP providers, I'm not sure if they swap their
IP's indeed,and send calls down my way with alternating IP's,
perhaps they're "smart" enough
Hi,
I can report, at least from my side, on not having issue with SRV
records, with Multiple SIP
providers. Did not have issues at least with the 1.8 branch. Are you
PBX's doing SRV lookups ?
Yes, I do have the srv_lookup parameter set to "yes".
What appears to be happening is that when the
ark 5
> == Using SIP RTP CoS mark 5
> -- Executing [200@local:1] Goto("SIP/588-", "fax-in,s,1")
> -- Goto (fax-in,s,1)
> -- Executing [s@fax-in:1] Answer("SIP/588-", "")
> -- Executing [s@fax-in:2] Wait("SIP/588-0000&qu
o("SIP/588-", "fax-in,s,1")
-- Goto (fax-in,s,1)
-- Executing [s@fax-in:1] Answer("SIP/588-", "")
-- Executing [s@fax-in:2] Wait("SIP/588-", "3")
-- Executing [s@fax-in:3] Set("SIP/588-&qu
Executing [s@fax-in:3] Set("SIP/588-",
"FAXFILE=/tmp/fax-588-20120312-092231.tiff")
-- Executing [s@fax-in:4] ReceiveFAX("SIP/588-",
"/tmp/fax-588-20120312-092231.tiff,f")
-- Channel 'SIP/588-' receiving FAX
&#
38 : 0
> Canceled : 0
> No FAX : 0
> Partial : 0
> Negotiation Failed : 0
> Train Failure : 0
> Protocol Error : 0
> IO Partial : 0
> IO Fail : 0
>
> Digium T.38
> Licensed Channels: 1
&g
Yes, I can give a higher ulimit, but I want to know why there were so much
fd's.
As I found out yesterday, the reason of running out of available file
descriptors was:
Some Agents in the Callcenter made ChanSpy on several Calls, but they
didn't stop spying with *-key, just hangup the phone, and ke
28 matches
Mail list logo
