Displaying 20 results from an estimated 2000 matches similar to: "SetCIDNum using SIP?"
2005 Jan 04
1
Newb howto request: *, Voice Pulse Connect, & SJPhone
I have been picking at Asterisk for about a week, and I think I'm close. I
was hoping for a little guidance to bring this on home.
I want to be able to make outgoing calls from my SJPhone clients using my
VoicePulse Connect account. I have the two requisite items from Voice Pulse,
but I've had no luck successfully integrating the VoicePulse settings into
iax.conf.
My current config:
2004 Aug 08
6
Voicepulse problems?
Is any one else having problems with Voicepulse today? Suddenly, I can't
register and calls to my Voicepulse numbers get a fast busy.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
2005 Jul 16
1
Voicepulse connect - unable to dial out, asterisk says "9696"
Hi,
for some weeks now I have been unable to make calls via my voicepulse
connect IAX account?
When I attempt the console looks like this:-
rt*CLI>
-- Executing Dial("SIP/2008-cf55",
"IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new
stack
-- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900
-- Call accepted by 66.234.228.160
2004 Dec 03
8
Why, why, why???
Help.
Why is it that I can call out from my GSBudgetone SIP phone but the
audio is "one-way'?
Why is it that when I call my asterisk phone number, I get a fast busy?
2007 Feb 26
2
SetCIDNum is not available on 1.4svn
Hello,
I am using the SetCIDNum dialplan application on 1.2 and 1.4.0; I've tried it
on 1.4svn 56126 and it does not recognise this application. Any idea?...
Thanks! __Yehavi:
2003 Dec 08
2
Problems with voicepulse.com
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound calling. Our
attempts at
IAX/IAX2 connectivity with VoicePulse have been less than successful.
We get
"Registration Refused" errors from Asterisk whenever we launch the
server. The
front-line support folks at VoicePulse suggested that we are
2004 May 25
3
Voice Pulse
Hello:
I am new to the list. I am trying to set up asterisk with voicepulse. I
have a voicepulse username + password, and SIP DID. When I login to
voicepulse, I have this under my devices tab:
Devices
*Login:* Sysxxxxxxx
*Password:* xxxxxxxxxx
*Context:* VPWS
*Connects to:* gw5.voicepulse.com
My question is: Do I need a 2.4.x kernel? Currently I am running
Debian/stable stock 2.2.x ? Has
2005 Jan 25
3
OT: pinout for"standard"telephoneheadsetrequired.?
> Many thanks Julian.
Are you looking for the pinout for a single plug 2.5mm (cellphone)
headset or a dual plug 3.5mm (computer) headset?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net
2005 Jan 06
3
IAX outgoing redundancy
Hello.
I am having an issue where sometimes the cheapest provider for certain
international destinations is not always reliable in completing calls.
However, there is not problem once the call is made (i.e. no lag or echo
or anything). The way I have it set up right now (for example) for Dar
es Salaam, Tanzania is:
exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1})
exten =>
2004 Sep 04
5
Wildcards and variable number of digits
Greetings,
I'm having a miserable time getting Asterisk working with FWD. All the
samples show something like...
exten => _7., ....
How do I get Asterisk to wait until the user is finished dialing instead of
trying as soon as it gets the second digit?
I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to
be able to dial others...
Same problem for outside
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks,
I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.
First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i'
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2004 Jul 20
10
Installing X100P
I attempted to install an X100P card but it was not correctly recognized
by my Redhat 9 install. I had a test install running without any cards
which was working great minus the outward dialing since no cards
existed. Now that I have a card, I want to add it to the system. Do I
have to scratch the whole current install in order to get the X100P
running on my system or is there a way to get it
2004 May 31
1
Failover: iconnecthere to voicepulse
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect
exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN}
Well,
2005 May 15
2
SIP Gerenal settings conufsion
I have a little confusion about the general settings (other than the
register values) in the SIP
General area. I understand that for examle in a SIP context like [FWD]
or [BROADVOICE]
the entries in those areas are ths settings that take effect in any
communication woth FWD and/or BROADVOICE. However, I'm confused as to
the purpose of the
"general" settings -- to what or which
2004 Nov 06
5
SIP Groups
I am wondering if there is a way to create a SIP/IAX group of outgoing
lines like Zap groups.
I am currently using the following method, but would like to use
features such as ?g2? that would list all the accounts for a SIP or IAX
connection.
exten =>
_1NXXNXXXXXX,1,Dial(SIP/account_name:Password@gw5.voicepulse.com/${EXTEN
})
exten =>
2004 Dec 22
1
Asterisk billing solution
Hello.
I am looking for a simple Asterisk billing solution. I expect about
50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all
IAX).
I need something that can handle monthly fees and per call charges
(depending on destination, obviously), and should provide a web
interface for customers and administrators.
Something that can tie in to one of the existing management GUIs
2004 Dec 18
5
Q about IAX (and IAXy)
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
server respends to your device's "apparent" IP and port (this
2016 Jul 30
3
Removing mailbox and password prompt for voicemail
If I remove the password, how can anyone access the mailbox if the
'mailbox' prompt is not played?
Nabeel
On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote:
> On Sat, 30 Jul 2016 06:43:47 +0100
> Nabeel <nabeelshikder at gmail.com> wrote:
> > I am using Asterisk voicemail on a CentOS 7 server. I would like to
> > be able to
2005 Jan 22
1
ASTCC: potential billing issue and "fix"
Before I start, I just want to say this is not necessarily a problem
with ASTCC, but may be a problem the way I have set up ASTCC (and
possibly the way others have set it up as well). The issue is that ASTCC
tries to match the pattern *anywhere* in the called number, not
necessarily only at the beginning.
I have set up ASTCC Routes like this:
1800 Tollfree Trunk1 0 0 100
1416 Canada Trunk2 0 0