Thanks A. J.
*José Pablo Méndez *
On Wed, Jan 22, 2014 at 3:22 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Wednesday 22 January 2014, José Pablo Méndez Soto wrote:
Hello,
Is there anyway to encrypt or scramble a bit the secret used to register
with a provider? Im talking
Hello,
Is there anyway to encrypt or scramble a bit the secret used to register
with a provider? Im talking about the
register = fromuser@fromdomain:secret@host
directive in
sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
This clever dude modified the code back in 1.4:
Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1):
http://packages.ubuntu.com/search?keywords=libspandspsearchon=namessuite=mavericksection=all
And having a sweet time with T.38 gateway. Oneiric already offers latest
pre18.
*José Pablo Méndez
*
On Wed, Jan 11,
I have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not
sufficient for running 1000 calls.
Only even ports will be used for RTP I think, odd ports are reserved for
RTCP, although I don't know how SIPp behaves in this line. 2000 ports
should be reduced to 1000 ports following my
Can you email me off list (since this isn't really Asterisk related and a
snom support issue, which I can help with) with some details and ideally a
SIP trace?
cheers,
Paul.
Closing this question with a final message including the [SOLVED] phrase
will definitely help the community I
Interestingly enough, they just list port 5060 in the Asterisk
interoperability guide:
http://wiki.snom.com/Interoperability/PBX/Asterisk
Could that mean it is a fixed setting? (crappy)
*José Pablo Méndez
*
On Fri, Jan 6, 2012 at 10:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Hello,
Reading through the Wiki:
Asterisk supports the ability to write dialplan instructions in the Lua
programming language. This method can be used as an alternative to or in
combination with extensions.conf and/or AEL. PBX lua allows users to use
the full power of lua to develop telephony
to build the gateway support?
Thanks again,
*José Pablo Méndez
*
On Thu, Jan 5, 2012 at 10:04 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote:
Hello,
Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/**wiki
Ah ok,
I got the incredible idea to go look into the make menuselect for a
res_fax_spandsp option after reading this:
http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html
I found it in 1.8, now you say it doesn't come with gateway support.
Thanks for the clarification
@sedwards.comwrote:
On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote:
My question is, what is the benefit of using Lua?
I've never used Lua, but I also have a curiosity about it.
A couple of years ago, I wrote my first dialplan in AEL. Some bits were
clumsy, minor syntax errors caused major
Ok so its not a cosmetic thing only. I eases your administration. Do a
point for performance.
Now, what about my questions regarding extending the systems caps by
building things asterisk could not build by itself. does it hold true?
On Jan 6, 2012 3:28 PM, Steve Edwards
kpflem...@digium.comwrote:
On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote:
Hello,
Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/**wiki/display/AST/T.38+Fax+**Gatewayhttps://wiki.asterisk.org/wiki/display/AST/T.38+Fax+GatewayI
wrote this
in my extensions.conf
Hello Bryant,
Have you seen the snom meetingpoint?
http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/
I don't own one, but it looks like a fine piece of hardware. And snom is
manufacturer of supported phones for Microsoft's Lync server (must say
something their quality
Hello,
Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in
my extensions.conf:
exten = 306,1,NoOp(Fax transmission)
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(DAHDI/3)-FXS port to fax machine
same
Can you show us how the previous INVITE Looked like vs the current one?
*José Pablo Méndez
*
On Sun, Jan 1, 2012 at 4:17 PM, cov...@ccs.covici.com wrote:
Hi. I am using asterisk 1.8 and everything was working fine when I was
at svn 342661. I then upgraded to vrsion 349339 and
May I ask why do you need different IP addresses to source calls? I mean,
its not a common practice, would like to understand the idea behind it.
*José Pablo Méndez
*
On Mon, Dec 19, 2011 at 11:07 PM, Anton Kvashenkin
anton.juga...@gmail.comwrote:
AFAIK you can add exterin= in
of the above?
Its a difficult question to ask/describe, so if I am not asking correctly
please let me know. Thanks a lot, really.
José Pablo Méndez
On Sun, Dec 18, 2011 at 12:18 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 12/18/2011 01:42 AM, José Pablo Méndez Soto wrote:
I have been
Thank you.
*José Pablo Méndez
*
On Sun, Dec 18, 2011 at 8:23 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 12/18/2011 01:22 PM, José Pablo Méndez Soto wrote:
Embarrassingly enough, I just tried the nat=no again both in the
general and peer sections and the blessed phone
Hey,
I have been testing with Cisco phones and have been able to register them
with new firmware 9.2.1 (7911/7945/7970). All worked until I realized that
from version 1.8.7.2, the VIA header contains the rport parameter, which
breaks the phone registration process. Basically, the device can´t
Hello,
I am trying to find a way to store the Register Call ID along with the peer
info, or at least extract it from a log. What can be tweak in chan_sip to
accomplish this? To illustrate, if the phone REGISTER message Call-ID
header was something like
Hello, I wen't through a lot of pain as well. Please try this script if you
can run your Asterisk installation on Ubuntu. The script is based on
Areski's own script.
Works flawlessly on server 10.10 and desktop 10.10 for me, but would like to
fix any possible bugs when used on different
I have been trying to install cdr-stats for a week now, but there is no
documentation worth the try and the amount of errors is huge. CUrrently
stuck running
python manage.py runserver 0.0.0.0:8000
I get
python manage.py runserver 0.0.0.0:8000
Error: No module named dilla
When starting apache,
Hi,
Trying to create templates that allow higher compression of sip.conf, so for
example:
[internal-number](!)
type=friend
secret=bigsecret
host=dynamic
context=internal
disallow=all
allow=ulaw
[100](internal-extensions)
mailbox=100@internal-extensions
[101](internal-extensions)
Yes sir,
We are pass the error. Works like a charm. I just documented this on our
new wiki:
http://voipcomsolutions.com/wiki/index.php?title=How_to_install_Asterisk_from_source_-_Google_Integration_ready
Thanks again
*José Pablo Méndez
*
2010/12/1 José Pablo Méndez Soto
Hello,
Can't get chan_gtalk.so module to load, neither res_jabber.so:
Asterisk*CLI module load chan_gtalk.so
Unable to load module chan_gtalk.so
Command 'module load chan_gtalk.so ' failed.
[Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error
loading module 'chan_gtalk.so':
Sorry never mind!
I got it to work after sof-linking to /lib/, and loading res_jabber.so
first, chan_gtalk.so second.
So in summary:
ln -s /usr/local/lib /lib/
asterisk-climodules load res_jabber.so
asterisk-climodules load chan_gtalk.so
Cheers!
*José Pablo Méndez
*
--
Hello,
We are working on implementing a solution for a medium service provider.
They were previously using a Cisco AS5300 gateway with some PRI trunks to
receive modem calls, then route them out the Internet.
The Telco they were buying the trunks to discovered this configuration and
restricted
if you want to.
Cary Fitch
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *José Pablo Méndez
Soto
*Sent:* Wednesday, November 24, 2010 7:31 PM
*To:* asterisk-users@lists.digium.com
*Subject
.
Cary
--
*From:* José Pablo Méndez Soto [mailto:aux...@gmail.com]
*Sent:* Wednesday, November 24, 2010 8:34 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* ca...@usawide.net
*Subject:* Re: [asterisk-users] Incoming calls through SS7
, and with MICA cards they have not a high resale
value,
so you'll probably end with them as paperweights unless you happen to have
some
stack of C549 cards to repurpose them.
Saludos,
H
On Wed, Nov 24, 2010 at 07:58:37PM -0600, José Pablo Méndez Soto wrote:
Hello,
We are working
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