21 March 2019 at 21:59:51, Darryl Moore wrote:
>
> > For a paging system? No you don't. A number of SNOM PA1's and a few
> > grandstream phones and you're golden.
>
> Are you suggesting using standard telephones (presumably in auto-answer
> speakerphone mode)
For a paging system? No you don't. A number of SNOM PA1's and a few
grandstream phones and you're golden. If you do need FXO or FXS, they are
just as easy to setup as well, and there are lots to choose from.
On Thu, Mar 21, 2019, 4:45 PM Ryan, Travis, wrote:
> You need more than an ATA. You
I've used SNOM PA1 before with good success.
On Thu, Mar 21, 2019, 2:59 PM Michael Munger, wrote:
> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>
>
> We’ve got a client with an old paging system that (supposedly) just takes
> an rj11 POTS connection, but
We do something like this, however we have two pairs of wires. One pair
is RS-485 for control running at 9600 baud. The other pair is baseband
audio which we control through relays on our intercoms. I can't imaging
trying to transmit digitally encoded audio over an RS485 network. There
are
I've seen this sort of thing where a DNS server is programmed in
resolv.conf but is not accessible over the network. Threads get blocked,
and you have to wait for the DNS query to timeout.
On 16-06-07 10:48 AM, Brent Davidson wrote:
I am having an issue with a couple of phones where they
Hey all. This isn't directly an Asterisk question, but it is Asterisk
related because I am using SIP on asterisk.
The last couple of days I found that our asterisk box was having all
packets originating from port 5060 being blocked.
If I moved my SIP port to any other port I could register
Ahh. Seen that before! That suggests to me that you don't have your
sip.conf records setup right.
What's your sip.conf look like?
On 15-05-28 04:51 PM, Luca Bertoncello wrote:
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
The phone you gave your wife is really old. Are you sure it
Bertoncello wrote:
Darryl Moore dar...@moores.ca schrieb:
Ahh. Seen that before! That suggests to me that you don't have your
sip.conf records setup right.
What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register = 004935:MYSECRET@pbxluca/004935
register
I'd start by turning on sip debugging in asterisk
sip set debug ip [your_phone_ip]
and use tcpdump or wireshark to see what the OS sees
tcpdump host [your_phone_ip] and udp port 5060
On 15-05-28 03:58 PM, Luca Bertoncello wrote:
Hi list!
I have a problem and I hope someone can help me...
What about this? No patches needed.
exten = my6003,1,Set(CALLERID(ALL)=MyCallerID)
same = n,Dial(SIP/6003@asterisk)
exten = 6003,n,Set(MyCallerID=test12345)
exten = 6003,n,Originate(local/my6003,app,meetme,6003,x)
On 14-12-25 06:46 AM, Anthony Messina wrote:
On Thursday, December 25, 2014
Hello list,
I have again come across a router which behaves very badly with my IAX2
packets. This time I've documented it and thought I'd share to see if
anyone else has seen similar issues.
I have two asterisk servers running behind a dlink DI-604 Internet
router. Both are trying to use the
On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
No such thing as 'free open source g729 license', if you actually read
the site:
There is regarding the copyright on the code. The fact it is also patent
encumbered is a different issue.
DISCLAIMER: You might have
Yup. That's what i do. The CLI version of linphone set to autoanswer, with
the audio jacks tied to our exernal sound system. Works well. The echo
cancellation in linphone helps a lot for speakerphones.
On Jan 16, 2014 7:51 AM, Administrator TOOTAI ad...@tootai.net wrote:
Hi list,
I have a
http://translate.google.com/translate_tts?tl=enq=i always find google
translate works well
http://translate.google.com/translate_tts?tl=frq=je trouve toujours google
translate fonctionne bien
On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote:
Hello,
Anyone know good quality text to
put it in a different context in your dial plan and use a gotoif statement
to control the times it is allowed to dial out. you can also redirect it to
a prerecorded message whenever someone tries to use it during the 'off'
time. no need for anything as brutal as disabling it in sip.conf.
On
patents, but I do look forward to g.729
entering the PD in 2016ish and joining MP3s which also recently became
an unencumbered format in most countries.
On Sun, 2013-10-06 at 00:13 +0800, Steve Underwood wrote:
On 10/05/2013 11:07 PM, Darryl Moore wrote:
[blink]
umm... they are software
On 2013-10-04 5:36 PM, Steve Underwood ste...@coppice.org wrote:
On 10/05/2013 01:32 AM, Darryl Moore wrote:
I'll explain.
The g.729 compression algorithm is not protected by copyright, though
specific instances may be. It is protected by a patent.
http://www.sipro.com/G-729.html
I'll explain.
The g.729 compression algorithm is not protected by copyright, though
specific instances may be. It is protected by a patent.
http://www.sipro.com/G-729.html
An open source version is available here:
http://asterisk.hosting.lv/
What stops you from using this, or even your own
Hi guys.
I would also add that in countries which do not recognize software
patents (New Zealand for example) there is no need to get a license and
the codecs can therefore be downloaded from http://asterisk.hosting.lv/
and used freely.
In countries where there is ambiguity about certain
OK list. Just in case anyone cares. I did figure out (sort of) why I was
not receiving any IAX packets.
http://lists.digium.com/pipermail/asterisk-users/2013-September/280590.html
My modules.conf file was the same one I had previously used in 1.8 and
it specifically loaded necessary modules
I saw this thread which is very similar to my issue, though I cannot
solve mine with iptables.
http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html
Using asterisk 11.5, IAX does not seem to be able to receive any
packets.
My IP tables looks like this:
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