Displaying 20 results from an estimated 400 matches similar to: "Opus codec in codecs.conf"
2010 Nov 02
0
Need testing: chan_unistim improvements
Hi All,
During last three month I have worked on improving functionality of Nortel
phones working with asterisk to replace existing Nortel station by asterisk.
Many improvments done, listed below. I have only i2002 phone and unable to
test if new version of channel correctly works with i2204 phone. If anyone
can test it and report issues, it would be great. Please visit mantis to
find out patch
2004 Oct 25
5
Nortel Phones.
Hello,
I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on
their asterisk implementation. My local dealer says they are not
compatible with any open source implementations. Is there a phone
compatibility list somewhere?
Cheers
Cian
2008 Nov 10
6
changing the size of voice packets
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because of bandwidth failure.
?
thanks in advance
Mani
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2005 Mar 07
3
UNISTIM channel driver available
Hello,
Cedric Hans has released an UNISTIM channel driver for asterisk (stable).
You can download it at :
http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2
Copy of README :
This is a channel driver for Unistim protocol. You can use at least Nortel
i2004 phones with it.
Only few features are supported : Send/Receive CallerID, Redial, SoftKeys,
SendText(), Music On Hold, Message Waiting
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2009 Jan 20
3
Forwarding calls and trasfer calls
Hi
How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions.
I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf.
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113
2003 Dec 02
7
Nortel i2004
Is anyone successfully using this phone with Asterisk? There is a lot
mentioned about CISCO but nothing about Nortel...
Alex.
2005 May 25
2
Nortel i2004 firmware upgrade.
I've been trying to look up information on upgrading firmware on a nortel
i2004 ip phone. I have this phone leftover from a trial, and it's
supposed to be upgradable to current firmwares. Since I also run a DMS I
was able to login to nortel's site and get all the firmware files, but All
the NTP's regarding firmware upgrading these are how to tell you BCM to
send the file to it.
2009 Feb 10
1
unistim and transfer calls
Hi
When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error?
Regards
/ralf
________________________________________________
Ralf Tr?skman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir:
2008 Feb 12
3
Nortel 1140E
Anyone get the Nortel 1140E phones working with Asterisk ?
These look like great phones and I would like to start using them on
our deployments. I know these will work with Asterisk but the sample
config files are hard to find. My next step, if I cant find anything on
this list is to purchase a Nortel Communication Server for testing.
If anyone has a used NCS that works with these phone via SIP
2020 Jun 11
1
OPUS encoded data size and bandwidth of encoder
Hey, I am having trouble with the size of the encoded bytes by Opus. I am
also having issue with the Bandwidth ctl.
Here is the scenario.
If I encode 16khz sampled audio:
opus_encoder_ctl(enc,OPUS_SET_MAX_BANDWIDTH(OPUS_BANDWIDTH_WIDEBAND))
opus_encoder_ctl(enc, OPUS_GET_BANDWIDTH(&x)) = 1102
opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&x)) = 1103
average encoded size = 120 bytes
if I
2014 May 29
1
Asterisk 11.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2007 Dec 20
7
ip phone suggestion for Asia?
Hi:
i am surveying ip phones for our company. we will use them with asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco and polycom are too expensive for us.
we try several china brand ip phones. they are all cheap and
some of them have good quality. but most of them won't offer future firmware
support, which we think it's important for ip phones.
2013 Jun 13
2
A quick question in terms of DAHDI channel
Hello,
I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command
connect*CLI> core show channeltypes
I would have response like:
connect*CLI> core show channeltypes
Type Description Devicestate Indications Transfer
---------- -----------
2008 Jan 16
2
Difference between TE121 and TE122
What's the difference between the TE121 and TE122. I read the description on
Digium's site and it isn't clear to me.
Best regards,
--
Guilherme Loch G?es
Visite nossa loja virtual: http://www.shopvoip.com.br
Not?cias e F?rum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
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2008 Nov 03
1
say load new
Hello all,
I would like to use say.conf settings but every time i restart
asterisk i have to load manualy "say load new" is there a way to do it
automaticaly i use asterisk 1.4.19
Thanks
2007 Dec 13
1
Sipura provisioning
Ok, I think I asked this previously but don't remember seeing an answer...
Yes, you can "tickle" an SPA94x or 962 and have it fetch a config from a
TFTP server... But is there no way to simply "push" a couple of lines
of XML config to it directly via an HTTP POST (sans TFTP server)?
Thanks,
-Philip
2008 Jan 16
1
Asterisk 1.4.17 and RXFAX via T38
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work with this?
2006 Jun 09
3
FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The
problem that I am having is that the VegaStream does not support incoming
registration from asterisk. VegaStream only allows outbound registration.
My question is does asterisk allow incoming registration from an FXO? If yes
how? Or better yet, has anybody been able to make the VegaStream FXO work
with asterisk? According
2018 Jan 27
2
Installation instructions for Opus are incorrect - maybe?
Before I got an log a ticket, can I just check I'm not doing anything wrong?
In 15.2, to install Opus:
1) run `make menuselect`
2) Highlight "Codec Translators" and press enter.
3) Scroll down to "codec_opus" in the section labeled "External"
4) Press enter to select the codec if it is not already selected.
... at this point, I see
XXX codec_opus
and a