Displaying 20 results from an estimated 20000 matches similar to: "Re: Will not authenticate incoming VOIP provider"
2006 Feb 06
1
Will not authenticate incoming VOIP provider calls
I running Asterisk 1.1 on Mandriva 2006.
Everything works fine, can connect with softphone, send outgoing calls to VOIP
provider.
The only (and big) problem is that Asterisk refuses to authenticate incoming
calls with the message (in the log):
Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129>
From what I've read in the various docs I could access, I
2008 Jul 11
0
Outgoing calls but no incoming calls with X100P
Hi all,
I have a problem with my asterisk box and an X100P FXO card. I am able to
place outgoing calls from my SIP phone (Cisco 7940) to any external number
using my PSTN line, but when I call my PSTN line from my cell phone, the
Cisco doesn't ring (and no message appears in the Asterisk CLI).
Here are my config files:
zaptel.conf
fxsks=1
loadzone = be
defaultzone = be
2010 Feb 20
1
Error redirecting an incoming call of a SIP provider to a local extension
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi all!
I am trying to redirect to a local extension the incoming calls that I
receive to an account which I have in iptel.org, but when receiving I'm
obtaining this error:
alderamin*CLI>
-- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820",
"SIP/300|30|tTrm") in new stack
[Feb 19 19:22:50] WARNING[19254]:
2010 Mar 16
1
Asterisk hangup all incoming calls after 10 seconds
Hello Gentleman,
I'm new to asterisk, this is my first instalation, first post... so I'd like
to apologize if this question has already been made. I googled but I
couldn't find nothing similar.
Here's the thing.
I'm migrating from ATA to Asterisk one of my client's office and I have a
very simple setup.
A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a
2011 Feb 21
2
calls are not going thru e1 line
I'm curious as to what versions of everything you are using. Reason
being this line " -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-00000000".
It states "DAHDI/i1/00256312261627-1..." and I don't recall seeing that
before (my 2.4.0 says " -- DAHDI/1-1 is proceeding passing it to
SIP/801-0000000c" [1-1 being the span and channel
2007 Feb 27
2
No sound with Playback() or Background()
After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange problem. There is no sound with Playback() or Background()
commands.
Even though, Asterisk console shows the file is being played when I call
the extension (i.e. echo test), I can't hear anything.
My echo test extension looks like this:
exten => 600,1,Answer
exten => 600,2,Playback(demo-echotest)
exten
2004 Jul 22
1
Faild Echotest
Hi
I have a cisco 7960 Phone that connects to my Asterisk server without a
problem.
But when I call the echotest it just hangs up, echotests from other VoIP
providers works just fine.
I have tried a softphone and it works just fine.
The error I get when the 7960 calls is this:
-- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack
-- Playing
2005 Oct 18
0
Slow dialling from PBX into * via E1
Hi :)
I have a little 'slow dialling' problem. When I dial, e.g.
200# for the Asterisk 'echo test' demo application from my PBX extension
1010, I see this in the console the instant I press the # key:
-- Starting simple switch on 'Zap/65-1'
-- Accepting overlap call from '1010' to '200' on channel 0/3, span 3
then exactly 3 seconds elapses, and
2006 Feb 01
1
SV: Re: CallerID Problem
This is what i found on Cisco's site:
"Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message.
Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message
2007 May 01
0
Re: [asterisk-dev] SRTP implementation
> Olle E Johansson wrote:
>>
>> 23 apr 2007 kl. 19.55 skrev Russell Bryant:
>>
>>> John Todd wrote:
>>>> To morph this into a -dev thread: if this patch were to become (again)
>>>> useful and error-free, is there any objection or usefulness in adding it
>>>> to TRUNK? Personally, I think there is, if there is a method by which
2009 Jun 18
2
Multiple Outgoing Lines: extensions.conf
Dear all,
I am currently trying to configure a PBX make use of a multiple of
outgoing lines, currently my extensions.conf looks something like below
>>
; extensions.conf
; 20th October 2008
[globals]
sip1=201
sip2=202
sip3=203
sip4=204
[general]
autofallthrough=yes
[default]
[incoming_calls]
exten => _89859715,1,Dial(SIP/201)
exten =>
2004 Aug 29
0
Asterisk H.323 channel...
Hi all,
I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2).
So far I have been using the H.323 channel included in the tarball (Nufone ?).
I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box :
=====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the
2005 Feb 01
0
No Sound Playback
New install, Calls are working phone to phone using gsm, ulaw or alaw
codec but when try and echo test or voicemail there is no playback.
I've tried turning on and off every codec and still no luck.
Asterisk says it's playing the sound file but I just don't hear
anything. I can't find any reason for this. I've tried the latest tar
and CVS with the same result.
2010 Oct 25
3
Extension Exists
Hi,
When a VOIP user dials an external number, the calls are routed through our SIP provider.
Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider?
Something like GotoIfExists(5551234 at incoming_calls)
Currently, I'm paying for all calls, regardless of whether they exist locally.
All DDIs exist in the incoming_calls context.
Thanks
Dan
2010 Jun 18
1
Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I have a playback function there.
But CLI reports:
CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542
2006 Feb 01
0
SV: Re: CallerID Problem
Seems to me like the negotiation fails for some reason. Maybe you are trying to use a callerid that isn't allowed?
Regards,
Jan
-----Ursprungligt meddelande-----
Fr?n: asterisk-users-bounces@lists.digium.com genom Gary Richardson
Skickat: on 2006-02-01 21:45
Till: Asterisk Users Mailing List - Non-Commercial Discussion
?mne: Re: [Asterisk-Users] Re: CallerID Problem
No, I'm not
2008 Feb 02
1
Echo() app doesn't work
Hello list,
New to asterisk and to the list (although experienced in Unix/Linux
administration).
Short problem description:
--------------------------
I cannot get the Echo() application to run on any 32bit platform I can get my
hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have
runs just fine. In all cases asterisk log shows the same -- that Echo() is
executed
Details:
2007 Sep 26
1
Routing issue
Hi list
I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk
solutions and appliances.
I installed TrixBox on a litle PC @ home and a x100p card which is
recognized as a Zaptel card, I made some in/outbound routes and they seem to
work but I have a problem with SIP softphones. I created 2 estensions 1000
and 1001 they're both in different cities, when I 1000
2005 May 12
0
FW: Incoming calls picked-up then simply hanged-up
Never mind. The Asterisk@Home documentation is incorrect the echotest is in
*43 and it works fine.
-----Original Message-----
From: fhunter [mailto:fhunter@survivorsoft.com]
Sent: Thursday, May 12, 2005 4:08 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Incoming calls picked-up then simply hanged-up
Something else I have noticed when
2004 Nov 24
0
No debugging informations on the CLI after patching with ast_data 1.0.2
Hi to everybody,
I have the problem that nearly no information are displayed on the
Asterisk CLI (asterisk -r). In former times (before patching Asterisk
1.0.2 with ast_data 1.0.2) it looks e.g. like this:
--- snip ---
-- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569
-- Accepting AUTHENTICATED call from 212.202.169.118, requested
format = 1024, actual format = 1024