Displaying 20 results from an estimated 10000 matches similar to: "Video in Skype for Asterisk"
2010 Apr 27
2
Problems for Skype for Asterisk
Is there an issue with running it with the latest from the 1.6.2 branch?
I did an svn update and make install and now when somebody comes in via
Skype, I get an infinite loop of:
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument
[Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument
[Apr 27 09:53:29] WARNING[10471]:
2010 Jul 16
2
SKYPE - Authenticate incoming call
>>
>>
>> Hi All,
>>
>> After getting licences for Skype for asterisk a while ago I finally
got
>> around to setting up a server with two channels and setting up a bcp
on
>> the skype end.
>>
>> The idea behind this is the following:
>>
>> Users can dial into the PBX, get authenticated and only after>
>>
2010 Jul 15
1
SKYPE - Authenticate incoming call automatically
Hi All,
After getting licences for Skype for asterisk a while ago I finally got
around to setting up a server with two channels and setting up a bcp on
the skype end.
The idea behind this is the following:
Users can dial into the PBX, get authenticated and only after
authentication get access to internal PBX extensions.
I CAN do this with a PIN, no sweat, but from a user perspective it
2011 Sep 13
1
High delay from Asterisk as PSTN simulator
I'm trying to use Asterisk as a PSTN simulator to run performance tests for
echo cancellation algorithms. I'm using the following configuration:
SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo()
Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan
application.
The problem is the high delay using this configuration: 20 ms only in
Asterisk 2.
2012 Jul 18
1
Asterisk 1.8.13 / res_fax / res_fax_digium
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP?
Also, the res_fax.conf.sample does not indicate v34 as a valid
2012 Feb 16
2
Asterisk && RTCP
Hello list,
I need to know about Asterisk's friendly nature with RTCP. I've phones
which support RTCP and they connect to the outer world via multiple
carriers. In one of my recent packet traces I've observed that the caller
initiated a call with rtcp string in SDP while for the same
call dialling our from Asterisk to the carrier has no RTCP string in SDP !
Can anyone please tell why
2010 Mar 12
1
Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.
I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine.
But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work.
I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and
2012 Jan 12
1
Questions on hardware or software-based echo cancellation
Hi,
I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ....) for
cases when a hardware ech canceller is present or not.
I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.
1. I'm correct thinking that it is then
2011 Nov 16
1
Server-to-server BLF
Hi all,
Do you have an idea on the best way on how to implement a system with
multiple Asterisk servers with BLF working in such a way that a peer on one
server can subscribe to another peer on the other server in a seamless
manner? Has anyone set-up a system like this before?
Thanks!
Regards,
Ronald
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2012 Jun 22
2
SIP over SSL TCP or SRTP?
Hello,
Which one of these ensures that SIP packets are sent and received in a
secure format so that users using public wifi don't allow MITM type of
attacks or others can't read the plaintext SIP packet info. VPN is not an
option. Looking for 2nd most secure to VPN.
P.S. Are both options part of the configs of Asterisk or need modules to be
selected and installed before doing the
2011 Jun 08
6
issues.asterisk.org/jira not working
Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>
2012 Jan 05
1
Where are the fax instructions?
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 => n,Hangup()
Call flow Im trying to pull out is as follows:
Zoiper -->
2010 Jul 25
1
Proprietary add-ons for Asterisk 1.8
At what stage will there be versions of the G.729 codec, res_cepstal,
skypeforasteric, Vestec, etc that'll work with 1.8? It would be good if
people using that software could participate in the Beta.
2010 Sep 15
1
Error loading skype_for_asterisk
This suddenly started appearing and I'm not sure why. Any ideas?
asterisk*CLI> module load chan_skype.so
Unable to load module chan_skype.so
Command 'module load chan_skype.so' failed.
[Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message
2012 May 10
3
Digium IP Phones
Hello,
Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features.
I cant find any website with opinions; any here? Are they really valuable
to the price? (D70 quite expensive)
Does the SDK for building apps is usable? Can you build powerfull apps?
Examples?
Many thanks
-------------- next part --------------
An HTML attachment was
2011 Nov 15
4
Multiple SIP endpoint registrations
Hi guys,
I want to ask if its possible to make calls using one SIP account,
The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center,
so the iPhone app will be using the same SIP account for all users
lets say for example:
iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)
2012 Mar 02
2
Digium FXS specifications and limits Question
Howdy All,
I'm considering Asterisk / Digium as a replacement to my existing phone
switch. I need to continue to be able to push analog lines between
multiple buildings in a campus environment.
The Digium Analog 410 Card manual states it's not recommended to go
beyond 1500 feet distance for an FXS card, and no line should leave the
building or be bundled. The 2400 Series Manual does
2012 Jul 19
1
Channel is rsrvd and does not turn off
Hi list.
I have Asterisk installed on a Debian 1.8 6 64-bit.
What happens is the following, some channels are not being hangup properly.
They run the hangup in dialplan, but the output of the command "core show
channels" shows several channels with status "rsrvd." Checking the server's
memory, the "top" command shows multiple processes and stopped using the
2010 May 21
2
Using unix socket to connect with database
Hello,
I am using asterisk realtime with a postgresql database on the same server.
In res_pgsql.conf I have specified
[general]
dbhost=localhost
dbport=5432
dbname=asteriskdb
dbuser=psql
dbsock=/tmp/.s.PGSQL.5432
Since both asterisk and db are on same server, I would like asterisk
to connect to db using the local unix socket. However asterisk is not
using the local unix socket to connect to
2012 Jan 16
2
How does Digium Repo install Dahdi on a virtual container while I can't do the same trying from source install?
Hello,
I can do simple, "yum install asterisk18-*" and it installs Asterisk and
Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and
smooth.
However, if I want to compile dahdi-linux on the same openvz then I get the
error, *"You do not appear to have the source for the 2.6.32-4-pve kernel
installed".*
*
*
1- Based on above error and Google search I have