Displaying 20 results from an estimated 1000 matches similar to: "Asterisk 1.8 wrong Def. Username"
2009 Aug 01
1
SNOM Phones Displays NR Frequently
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code: snom190-SIP 3.56m
snom320-SIP - snom320 jffs2 v3.36
snom300-SIP - snom300-SIP 6.5.2
Asterisk version - Asterisk
2010 Dec 22
0
Asterisk 1.8.1.1 Multiple Parking Lots
Asterisk Version: 1.8.1.1
Problem: Multiple Parking Lots
Issue: Not redirecting to the right parking lot. Always uses the first
parking lot from "parkedcalls show" or "features show"
Asterisk Working Version: 1.6.1
Steps Taken:
In features.conf added:
[parkinglot_test]
context => parkedcalls-test
parkext => 700
parkpos => 701-710
parkingtime => 120
findslot
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
2013 May 21
1
Failed to authenticate device "Ext 110"
I'm having a strange problem recently with a Yealink SIP-T28P phone
connected to Asterisk 11.4.0 via openvpn. It was working fine for months,
and now when I dial anything from the phone, it shows "Forbidden", and the
Asterisk console shows:
[May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189
handle_request_invite: Failed to authenticate device "Ext 110" <
2011 Mar 06
1
Early codec selection / negotiation
Hi,
This seems to be a fairly common question, but I have Googled for this quite
a bit and looked at the Asterisk documentation/book and haven't been able to
find an answer.
My question is:
Can I get my IP phone to select a different codec depending on the final
destination of each call?
I've got these things connected to my Asterisk box:
- Snom 300 phone (supports g729 and
2015 Mar 26
2
call between snom 300 and aastra 6731i
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i 2.6.0.2019
i have configured the trunks like below
100 in snom300
200 in snom300
300 in aastra6731i
400 in x-lite
the calls between x-lite and aastra ====ok inbound and outbound
the calls between x-lite and snom300====> ok inbound and
2007 Mar 14
1
strange things on call transfer
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Hi,
I'm setting up an Asterisk system which is connected to an Alcatel 4400
PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a
call by hitting the # key, I get this messages and nothing happens on
the phone:
WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame
that isn't a multiple of 50 bytes long from
2015 Mar 27
0
call between snom 300 and aastra 6731i
please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>:
> hello list
>
> i need your help please regarding an issue with snom300 and aastra6731i
> using asterisk
>
> 11.13.0 asterisk
>
> snom 300 8.7.3.25
>
> astra 6731i 2.6.0.2019
>
> i have configured the trunks like
2009 Jan 30
2
Backup to spare drive (rsync / crontab)
I am using rsync and crontab to perform scheduled backups on FreeBSD AMD64 Rel. 7.0
I am following process described here for rsync :
http://samba.anu.edu.au/rsync/examples.html
I have a backup script's created for daily, weekly, monthly.
This is one example - the daily (/backup is a seperate physical drive) :
#daily backup script
rsync -a --delete /usr/home/data/Access/
2010 Mar 02
0
1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?
Asterisk 1.4.29
BLF-SUBSCRIBE go to internal IP (ngrep output):
U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060
SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From:
<sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To:
<sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:
2007 Nov 03
0
OT: Snom 300 losing config?
Hi,
I've had a Snom 300 connected to my Asterisk box at home for 12 months
or so now. Recently it lost all its settings and I had to reconfigure
it via the built in website.
For a few weeks it was fine. Couple of days ago it lost its settings again.
I logged in to its web server and thought I would upgrade the
firmware. It seems to be running an old version:
Phone Type: snom300-SIP
2007 Jul 12
0
No subject
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to A
realm=192.168.0.2
context = default ;Default for incoming calls
[5549]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
type=friend ;(inbound and outbound calls accepted)
secret=localphone ; obvious password for testing
host=dynamic
callerid=Jason White <5549>
dtmfmode=auto
mailbox=5549 ;(Asterisk VM-system's
2015 Mar 27
2
call between snom 300 and aastra 6731i
You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which
codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
> please no body has som with aastra can help me in this issue
>
> 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit
>
2006 May 04
1
Unwanted conference with snom320 and asterisk 1.07bristuffed
Under Advanced make sure this is set:
Call join on Xfer (2 calls): to off
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tommaso
Calosi
Sent: Thursday, May 04, 2006 4:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unwanted conference with snom320 and asterisk
2010 Oct 28
0
Intermittent failure when placing calls - unable to create channel of type SIP
Hello community,
I've been running Asterisk on an embedded device for about six months, and
my operation has been largely trouble-free. I'm hoping I could get some help
with a minor problem:
Every week or three, my PBX gets stuck in a state where it can receive
calls, but it becomes completely unable to originate outgoing calls until I
do a "sip reload". After doing the SIP
2015 Mar 27
0
call between snom 300 and aastra 6731i
thank you for your response below the asterisk -vvvr
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192",
"user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192",
"TOUCH_MONITOR=1427481319.470") in new stack
--
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
2015 May 31
2
Signaling incoming call
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Guenther Boelter <gboelter at gmail.com> schrieb:
> -----BEGIN PGP SIGNED MESSAGE-----
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>
> On 05/31/2015 02:31 PM, Luca Bertoncello wrote:
> > Hi list!
> >
> > Now all works as expected, at least in the simulation I did with
> > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
2008 May 21
1
using gtalk received instant messages in the dialplan
I have been doing some reading about gtalk and asterisk. Most of it is
pointed to enable using gtalk for making phonecalls. Would it be
possible to use gtalk instant messaging input (just some text send to
the gtalk account configured on an asterisk box) into the dialplan.
This way you could use gtalk im to trigger all kind of events like
sending an sms, adding sip entries to the system,
2014 Feb 18
1
Syntax error for Realtime SQLite3
I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While
everything seems to be working fine I keep getting this error on my log
files:
[2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not
execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" =
'5060', "regseconds" = '1392692118',