Displaying 20 results from an estimated 3000 matches similar to: "One way voice with Asterisk"
2006 Dec 18
0
pap2/wrt54gs/asterisk
I am having trouble setting this system up and wonder if some one help me.
Does anyone know what is missing if anything to get 2 phones on my
asterisk home server to be able to call each other.
I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2
extensions 5060/5061, this is on the lan side of my gateway/router
WRT54G 192.168.1.1
BusyBox v1.00 (2006.11.07-01:40+0000)
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;-----------------------------------------
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER
2009 Jul 14
1
Polycom Spectralink 8002 WiFi Phones
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go thru in second line, but as soon as you terminate both
calls it wont recieve any calls again.
Heres
2020 Jun 13
0
Voice "broken" during calls
So the call used Alaw as Codec.
> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>:
>
> 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 :
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
2020 Jun 13
3
Voice "broken" during calls
Am 13.06.2020 09:30, schrieb Luca Bertoncello:
Hi again (again)
I noticed right now another strange detail...
I made a call using my mobile phone (connected to the Asterisk). The
quality was top...
Maybe is the problem in a codec used from our phones at homes?
Could someone suggest me how to check the codec used by my mobile phone
and the codec used by the phones at home?
Thanks
Luca
2009 Jul 15
4
DEVICE_STATE() and Asterisk 1.6.0.10
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than "NOT_INUSE".
I have two extensions: 6666 and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I then use 6666 to call
6668 and in the dialplan have a noop to see what
2007 May 13
0
Asterisknow b5 - trouble registering at voip provider
Hi, there.
I have asterisknow beta 5 with the following data:
Ip 192.168.0.60
mask 255.255.255.0
gw 192.168.0.1
the router (a linksys) has port forwarded the port udp 5060 and from
16384 to 16482 udp-tcp from the internet to the asterisk machine.
the only protocol allowed is g729. Which work fine for the ip phones I
already have setup in the LAN.
My problem is trying to register to a voip
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
What's the difference between user "123" and "devries"? Based on the
output here, they seem the same..?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
123
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
2008 Nov 12
1
What are the minimum realtime fields for sipusers?
I'm trying to get sipusers working with a realtime odbc database on
Asterisk 1.6. We have sippeers working from the database, but need
sipusers to be in a separate table for other implementation reasons.
sip show user test load returns results from the database.
CLI> sip show user test load
* Name : test
Secret : <Set>
MD5Secret : <Not set>
2014 Jun 25
2
OPTIONS Request without username <-> Forbidden
Hi gurus!!!
I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn
Every minute asterisk sends an OPTION Request, i beleived that it's related
to qualify functions.
The every minute annoyng answer of the pstn is "403 Forbidden".
Some people told that asterisk is not sending the username in the OPTION,
required by the pstn.
Taking a look of the example of rfc3261.txt
2009 Oct 18
1
Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
defaultuser= ;; SIP-ID
fromuser= ;;SIP-ID
context=sipgate_in
fromdomain=sipgate.com
host=sipgate.com
2015 Feb 20
0
sipsak 200 for a user, but 404 for a different user...why?
On 2/20/15 6:15 AM, thufir wrote:
> What's the difference between user "123" and "devries"? Based on the
> output here, they seem the same..?
>
> tleilax*CLI>
> tleilax*CLI> sip show users
> Username Secret Accountcode
> Def.Context ACL Forcerport
> 201 password 201
> default
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
> I suppose that you enable the video support on sip.conf, right?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>
2016 Jan 21
2
NAME/USERNAME conflict
Hi.
we are experimenting a strange issue in our PBX.
By example: if we dial to the 100, the call is answered in 199. We dont
have any redirection for that, but the cli show the same issue when request
show peers. Aditionally, the user 100 use the ip address 192.168.11.100,
and the cli show connected the user from 192.168.11.160 (that ip is
assigned to the user 199)
PBX*CLI> sip show peers
2011 Mar 02
1
Asterisk 1.8 SIP realtime and NAT
Hi
After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime cache.
For example
* Name : 501
Realtime peer: Yes, cached
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : pack-local
Subscr.Cont. : <Not set>
Language :
AMA flags :
2015 Mar 03
6
TLS, SRTP, Asterisk11 and Snom870s
CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5
I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else. At the moment I am trying to
get TLS functioning with our Snom870 desk-sets. And I am not having
much luck.
Since this is an extraordinarily (to me) Byzantine environemnt I am
going to ask if any of you have gotten