Displaying 20 results from an estimated 7000 matches similar to: "integrated voip originator, to digitize audio once and only once?"
2003 Nov 09
1
Dialing 800 numbers through FWD or SIPphone?
Hi,
Does anyone know how to dial toll-free (800) numbers through FWD or Siphone?
Using the configuration below, I can dial out to SIPphone.com users by
simply
dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing
1383<FWD#>
However, when I dial 18005551212 through SIPphone, or through FWD (depending
upon which line is selected in "; 800 Toll Free Numbers"
2003 Nov 18
3
"Unable to find path from G729A to ULAW" on Sipphone.com
I seem to be having a problem with transcoding and/or agreeing on a
valid codec. I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.
Here's what I see in the console:
NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1259545280]: File
2004 Feb 03
1
sipphone dialing out problem
Hello
when i dial a toll free no using sipphone i get this error message. How do i solve this?
Any help will be appreciated.
console message:
Starting simple switch on 'Zap/2-1'
-- Executing SetCallerID("Zap/2-1", "17473863282") in new stack
-- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack
-- Executing
2006 Apr 26
1
getting asterisk to reliably answer a voip line
I have a sipphone.com account, with asterisk set to
answer incoming calls, using the following settings
(phone number and password omitted) in the Peer
Details for the SIP Trunk:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
fromdomain=proxy01.sipphone.com
fromuser=1747xxxxxxx
host=proxy01.sipphone.com
insecure=very
secret=xxxxx
type=peer
username=1747xxxxxxx
The Asterisk machine is
2004 Feb 03
2
IPKall->FWD->Asterisk
Hi Folks,
I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing
2006 Apr 21
2
confused about iax and voip providers termination
Hey guys,
I'm actively trying to get the "big" picture on how all this works and
relates to each other.
I've gone through some basic examples from the book and from the sample
files just fine.
Now, I've setup an account with a VOIP provider which does IAX termination
(exgn.net)
After getting an account and following their steps, I can make calls out
using my IAX (cubix) and
2005 Mar 21
3
US pstn => voip
Hi
I believe this is due to the way US phone systems work, however I'm going to
ask anyway. In the UK there are several providers who provide national rate
PSTN => Voip gateways which are free to receive calls on, (for the
recipient), the caller pays the cost of calling. E.g 0844 0870 etc.
I am looking for a US provier who offers the same sort of system. I don't
call the US but I
2006 Nov 16
0
jitterbuffer in pure voip (sip/iax) - what is best practice
I know, that jitterbuffer should be set at receiving side and on
outgoing call leg,
ie. if sipphone calls to asterisk and outgoing to zap chanel, I should
set jitterbuffer on zap channel (to dejjitter audio stream from sipphone)
but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)?
I have following setup (homeworkers using sip phone connected to home
asterisk via SIP and
2004 Apr 05
0
SingTel ready to break into web telephony
http://www.smh.com.au/articles/2004/04/05/1081017104255.html
SingTel ready to break into web telephony
April 6, 2004
Singapore Telecommunications is teaming up with US internet phone
start-up SIPphone to offer low cost, and in some cases free, phone
services over the web.
The deal, expected to be announced today, will allow SIPphone - started
by MP3.com founder
2004 Nov 22
1
Using IPKall and SIP with insecure=very
Hi all,
I've got one of those cool free incoming IPKall phone numbers from
www.ipkall.com. These numbers just connect to the SIP proxy of your
choice, they default to Frreworld Dialup. You can use them with your own
sip proxy on asterisk. My config for this is below.
The trouble I'm having is the incoming calls do not seem to hit the
section in sip.conf for the call. With sip
2006 May 01
1
Using frequent keepalives to eliminate need for NAT port forwarding?
I have an asterisk system behind NAT, and need to
connect to public PSTN originators via SIP or IAX2,
but don't have the option of forwarding any ports
(4569, 5060, etc) to the asterisk system. However, the
NAT system does properly establish transient UDP
forwarding on the basis of outgoing connections, so is
it possible to configure asterisk to send frequent
keepalive UDP packets (say every
2009 Sep 15
0
1.6.2.0-rc1 intermittent voicemail problem ?
1.6.2.0-rc1. I am having trouble with voice mail intermittently not
working correctly on CHANUNAVAIL. (it may happen for other statuses
too, haven't checked). Basically here's what happens:
-- Executing [1651xxxxxx at mydids:1]
Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack
-- Executing [s at macro-phone:1]
2006 Feb 27
2
jitterbuffer and DTMF conflict?
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too: http://bugs.digium.com/view.php?id=6011
Can anyone suggest a workaround (other than jitterbuffer=off)?
- Mike
2007 Jan 30
3
Toll-free dialing via PRI problem
We have a PRI from Telepacific. Asterisk 1.2 and a Sangoma A101 T1 card.
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but
the calls are never answered. All other calls, and most toll-free numbers
are not affected. The numbers that are affected are all travel related
companies (United Airlines, American Airlines, US Air, Starwood Hotels,
etc.) we cannot connect to
2003 Aug 21
0
Dial in modem speeds over VoIP?
Hi all,
I like to have a dial in modem on my toll free number so that
when I or my employees travel, they can always get in for net
access to read email if no better method is available.
Right now, my Panasonic KX-TD1232D PBX receives the call on a
POTS line and routes it to an analog modem. The speeds achieved
are around 24 kbps due to the digitization of the KX-TD1232D
introducing
2003 Sep 29
1
Needed: Configuration Examples for VoIP Providers Asterisk can Register With
Hi all,
I would like people to email me at 'leif at hacklocalhost dot com' some
example configuration files for VoIP providers which * can register
with. I am going to expand upon the FWD php "wizard" I created for
these other providers, but I need some examples as I don't actually use
anything but IAXtel and FWD.
So far sipphone and iaxtel has been mentioned. I can
2009 Jan 13
1
FWD and IPCall
I tried this
http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html
But I am NOT getting call in asterisk.
SIP.conf file :
_________________
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
externhost=59.160.44.21
localnet=192.168.0.2/255.255.255.0
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test at 10.10.10.16:5060
;
2007 Oct 04
0
Friday VOIP Users Conference 12:30 PM EDT
This Friday at 12:30 PM EDT
We hope to hear more about Astricon and the 1.6 version. A UK legal
professional, John Halton of Cripps Harries Hall LLP, joins us to
discuss how the law is coming to terms with VOIP. We also expect a
visit from Arick of IPKall about what's cooking with them. Most of
all, we expect you, the community to share in this experience!
For more info see:
2006 Dec 20
13
Need quality toll free 800 number over IAX?
Hi List
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Thanks
--
Chris Blunt
Entropy IT Ltd
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm
2009 Sep 19
0
IPKall using iax
Is it possible to receive a call via IPKall through IAX connectivity without registration?
If so how to set it up.
I've run-into and old link;
http://forum.voxilla.com/ipkall-support-forum/ipkall-beta-testing-iax-connectivity-without-registration-26728.html
--
Joseph