Displaying 20 results from an estimated 600 matches similar to: "VIDEO ON 1.0.7 stable"
2005 May 26
5
SIP Soft Video phone for Asterisk usage
I am looking for a SIP Soft Video phone, which I can use with Asterisk.
If you have one installed (regardless if free or purchased) please tell
me which one, the settings in Asterisk and your experience with it.
bye
Ronald
2005 Jun 03
3
4 port BRI options ?
Hi there
I am in the UK.. and am running latest asterisk on FC1 (2.4 kernel). I would
like to know what the best option is for a 4 port BRI card. I notice Digium
don't provide one.. I have heard the Junghanns do one...but are there others
??
Is the Junghanns card reliable/stable with good sound quality ?? I notice it
is very expensive in a per port comparison with the Digium cards.... hence
2004 Jun 16
2
embedded Asterisk
Hi All,
I have a thin cliente here that i want to run asterisk:
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip
- NS Cx5530a Southbridge National Semiconductors SC2200
- NS PC97317 in chipset
- 32MB Compact Flash
- 64MB Ram
- 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816
Some tip?
I have a ide>flash
2004 Jan 16
1
ERROR[8192]
Hi all!
I get this error when trying to start asterisk:
ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk
What can be the problem?
Thank you!
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
UK +44 870 - 3403539
FWD 64662
sip:ipfone@sipserver.com.br
www.ipfone.com.br
info@ipfone.com.br
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2003 Sep 23
2
error message playing .mp3
> -----Original Message-----
> From: listas iPfone [mailto:listas@ipfone.com.br]
>
> Somebody knows why asterisk gives me that error wile playing .mp3
files?
>
> The files play well but the message aperas any way:
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0
of
> 4
> bytes) (No such file or directory)!
Listas,
You might try down-sampling
2020 Aug 07
2
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
I'm trying to transition from AMI to ARI.
Running into a small hiccup when I try to create (originate a call) with the caller id name and number
I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application.
curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at
2020 Aug 07
3
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Dan,
as far as PPI and PAI Header, we use the channel Vars in order to do that.
In Latest Asterisk you can set Channel vars within the create command in
the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran
On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote:
> An
2020 Aug 06
2
Is it possible to use Stasis to control both legs of a Local channel created using ARI?
I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis.
I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter.
[mycontext]
exten =>
2020 Aug 10
2
With ARI, is it possible to create (originate) a call and pass both the caller id name and number?
Hi Dan,
i would do something like this (it is not a copy of what we are doing but
an example of how i would do it)
Important here is the "--data" and "-H" Option as well as the "variables"
section within the Body. I added the callerid function here as well as it
is the sample in the asterisk wiki.
curl -v -H "Content-Type: application/json" -u
2004 Apr 15
1
sip videosupport
Hi all
I was tryed to connect to mysip.ch scs_client by siemens that isn't
works well.
Does anyones knows to work H/W or S/W applictations in asterisk SIP
videosupport?
Regards
mack_jpn
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2004 Dec 07
1
H.323 trunking
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the
2011 Jul 13
1
Connect Avaya to Asterisk PBX
Hi List,
I have another issue on allowing outgoing calls to PSTN on Asterisk via
Avaya Phones, I hope that anyone could help me fix this issue:
*When I dial through Avaya phone i just here a "good bye message" reply
from asterisk server. And here is the log:*
== Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling
back to exten 's'
== Starting
2009 Aug 14
2
no ring tone
how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your external domain name here)
nat=yes
qualify=yes
canreinvite=no
Add extra codecs to
2003 Dec 08
2
snom X MOH
Hi all!
I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension).
Someone with that problem?
I downgrade to 2.01s but nothing changes.
Miklos
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2003 Apr 01
1
Problems Calling Toll-free number
After a long working evening yesterday, now my * box place and receive calls
with H323,SIP and ISDN line.
Calling from the office to an outside line, happens:
- If I call a mobile number and the called answers, all goes ok
- If I call a number at home/office, and it's answered , all goes ok
- If I call a toll-free number with an IVR system, nothing happens: it
continues to ring indefinitely
2003 Oct 31
2
MOH problem
Hi all!
Every time i receive a sip call MOH begin to play and i can?t talk to the caller.
My setup is the default.
Someone knows what is the problem?
thanks
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
FWD 64662
ICH 31451543
www.ipfone.com.br
info@ipfone.com.br
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2005 Jul 11
4
Video phone settings???
I have three video phones here for testing:
Extension 6003 is Eyebeam
Extension 6004 is a hard phone (model 8770)
Extension 6005 is a hard phone (model 8882)
Can anybody have a look at my settings and the output I get from all
kinds of dialings, please.
The sip settings for all phones is (user / password different):
[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
2005 Sep 28
1
Asterisk sound files, audio bandwidth, and sound quality
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a telephone call, but for
anything else, it leaves something to be desired.
Case in point -- if you compare the