Displaying 20 results from an estimated 500 matches similar to: "GotoIf with Authenticate"
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2005 Feb 15
0
oh323 question
I'm trying to connect an asterisk server via oh323 to a Lucent iMerge.
I patched the code due so that Lucent can handle asterisk's ver4 h323
http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration
I can now successfully dial in to our company over multiple lines.
The issue is when I dial out
The first outgoing call connects to an outside user A
The second call drops the first
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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2005 Aug 10
0
tdm400p / outbound zap prob
I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p
w/ 4 FXO. Incoming calls work fine, outbound I get this:
-- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack
-- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack
-- Executing GotoIf("SIP/231-af2b", "0?19") in new stack
2005 Sep 14
2
PRI to PRI passthrough with DID intact
I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink
T1 ---- Asterisk.
I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk
extensions over the T1.
I do not get DID nor CID on the Asterisk, so I want to use PRI between the
PBXs.
I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are
different cards)
I see this as my least
2005 Feb 27
5
Outbound call on TDM400P
Ok all here is a strange one......
I have a TDM400P with 3 fxo modules. I can very rarely make an outbound
call to the PSTN....about once every 10 tries. However if I use a
analog phone pluged into the same phone line as one of the tdm channels
say channel 4, and I place the analog phone off hook and then place a
call via asterisk , it work everytime. It seems like the TDM400p is
having
2018 Jul 30
0
2.3.2.1 - EC keys suppport?
> I did some local testing and it seems that you are using a curve that is not acceptable for openssl as a server key.
>
> I tested with openssl s_server -cert ec-cert.pem -key ec-key.pem -port 5555
>
> using cert generated with brainpool. Everything works if I use prime256v1 or secp521r1. This is a limitation in OpenSSL and not something we can really do anything about.
>
>
2018 Jul 30
3
2.3.2.1 - EC keys suppport?
> On 30 July 2018 at 20:37 ????? <vtol at gmx.net> wrote:
>
>
>
> >>>>>>> facing [ no shared cipher ] error with EC private keys.
> >>>>>> the client connecting to your instance has to support ecdsa
> >>>>>>
> >>>>>>
> >>>>> It does - Thunderbird 60.0b10 (64-bit)
>
2018 Jul 30
2
2.3.2.1 - EC keys suppport?
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On 30 July 2018 at 21:00 ѽ҉ᶬḳ℠ <
<a href="mailto:vtol@gmx.net">vtol@gmx.net</a>> wrote:
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2005 Jul 18
6
Panasonic KX-TD500
Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500.
I have Asterisk connected via crossover to the TE110P. We are able to
make internal calls into the Asterisk Box but the PBX vendor (I know
nothing about the KX-TD500) tells us it is not possible route DID over
the trunk. I find this hard to believe. Anyone have any luck with this?
Thanks!
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1998 Sep 03
3
Dual personality samba server
Dear All,
I've just installed Samba version 1.9.18p10 on one of our HP-UX boxes.
The unix box is advertising itself with two WINS names.
netbios name = ukswi0103
netbios aliases = ukswi0104
in smb.conf
I've been playing with the idea of changing the behaviour of the server
based on what the client calls it. What I'd like to do is have samba do
security = user
if the client
2015 Mar 25
2
TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same provider in my asterisk
i can call all numbers just the numbers are configured in thses ip phones.
but when i configured the same trunk in x-lite i can call theses ip-phones
without
2005 Aug 22
2
TE110P problem
Im using a TE110P as a trunk to a Panasonic KD-500 everything works
well.....but Im having this problem where one of the channels becomes
blocked with a partial phone number after about two days. So if the
channel that becomes blocked is channels 23 no calls can get in. If the
channel that gets blocked is 22 then one call can get in. The only way
to clear this is to reboot the server.
This
2008 Oct 28
1
Multiline Analog Setup
What is involved in provisioning Asterisk to use a multiline analog service from our local telco?
I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk
interprets and deals with two incoming calls and/or two outgoing calls?
Thanks!
jlc
2005 Jan 09
2
ASTCC Trunk and Routes Configuration
Dear List members-
I am trying to configure ASTCC (Asterisk calling card application) but
having a hard time to configure it properly. My project deadline is
approaching and couldn't figure out how to make ASTCC functional. Here
are some details what I have done so far.
1) I have installed ASTCC successfully.
2) I can access astcc-admin.cgi script without any problem.
3) I have created
2004 Jun 10
4
How to get the Called id with AGI
Hi all,
Is there a way to get the "called id" (the B number) with AGI perl ?
I know how to get the caller id which is working fine and is just below:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$callerid = $input{'callerid'};
$AGI->say_digits($callerid);
}
Thanks in advance,
Angel.
2011 Sep 28
2
PSTN connectivity
Hi All,
I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.
1. OUTBOUND caller id and Dialing rules in Freepbx.
2. INBOUND route
When i call to the PSTN number before
2005 Jan 28
4
FW: FAQ missing info? Asterisk@home V 0.4
Just installed V 0.4 of asterisk@home
Programmed up 3 sip budgetone extensions, they call call each other
fine.
Tried to dial '9' for an outside line through an X100P to a packet8 ATA
but got 'all circuits are busy now'.
Here is the console output.
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/30-8d25'
-- Executing
2015 Mar 20
3
outbound calls
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i configure
the trunk directly in x-lite i can contact this number without issue
below the cli
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0149xxxxxx at
2005 Feb 24
2
Delay after entering digits with IVR
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit