Displaying 20 results from an estimated 1000 matches similar to: "Vote on whether SipPhone should support ISN routing."
2009 Feb 17
0
Optimizing this script for calling Skype users from Asterisk
I have written this configuration script which uses OpenSky to make Skype
calls directly from Asterisk devices using my companies SIP to Skype
gateway. Users can dial skype_anyskypeusername or manually add names or
extensions which can get mapped to the correct dialing sequence. The right
sequence is username at opensky.gizmo5.com but that gets mapped to sipphone
address so I set that up to map
2009 Aug 05
0
Asterisk with gizmo5 and google voice only takes one call at a time.
my problem is this. I have google forward the call to gizmo5. I have this line in my sip file :
register => user:password at proxy01.sipphone.com
I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it?
At the end of my sip file i have this
[Calls-From-Gizmo-Network]
type=user
context=demo
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
2009 Mar 19
0
Can I tell if a call picked up on PSTN extension... for example?
Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-)
My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL).
Calls come in and are
2005 Feb 09
0
Asterisk and SIPphone won't cooperate
When attempting to call one of the example numbers, like 17474745000, I
only get "488 Not Acceptable Here". It works fine when I configure the
softphone (Xten X-Lite) to use sipphone's server directly. Am I missing
something? Here's my relevant config sections:
sip.conf:
in [general]:
register => 17472442457:mypassword@proxy01.sipphone.com
[sipphone]
type=friend
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"
2005 Jan 03
0
Re: Asterisk won't register with sipphone.com
Hello All.
I started setting up my Asterisk system yesterday and everything was going
well, i have registered with sipphone.com and set-up my Asterisk system to
register with sipphone per the sip.conf file below.
It was registered perfectly but I could not receive calls so I added in the
line "insecure-very" and I then used the Washington DC access number to test
and the phone
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
2004 May 18
2
registering in sipphone
for inbound calls, i can register
context = from-sipphone
register => 1747xxxxxxx:passwd@proxy01.sipphone.com
but how do i configure to make outbound calls to them?
exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1)
....
[dial-sipphone]
;
; SIP to sipphone.com
;
exten => _X.,1,Dial(SIP/${EXTEN}@??????)
^^^^^^
2005 Mar 03
2
FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls
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 Apr 26
1
Problems registering with Sipphone
Has anyone else had problems registering with Sipphone over the last
few weeks?
Previously, this had worked fine. I contacted Sipphone technical
support, but they're not much help.
register => 17471234567:password@northamerica.sipphone.com/123
2004 Oct 05
2
SIPphone All-in-One: coments anyone?
Hello,
can anyone comment on how one could use SIPphone's $89 All-in-One adapter
with Asterisk? Sounds to me like it should work as both a FXO and FXS.
It would be a cheap way of getting started with Asterisk and PSTN.
Any comments on the SIPphone FX200?
Any comments on SIPphone in general?
Thank you for your help
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:
[sipphone]
type=peer
2003 Aug 12
1
Working with FWD, IPTel, SIPPhone?
I'll admit it. I'm a asterisk newbie (but no stranger to telephony).
The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone
specials) on a private segment calling to a Linux box acting as the
segment's firewall with a leg on our public network. The phones are
setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks
to the Asterisk HOWTO).
Getting IAX
2004 Oct 05
1
asterisk with sipphone.com
Hi all.
I found a connection error from sipphone.com.
It seems 'realm based authentication' by sipphone.com.
any ideas?
Regards.
mack
2005 Aug 08
0
OT: Anyone having issues with sipphone?
All of a sudden, my account doesn't appear to work, or even perhaps
exist with SIPPhone. Is anyone else having trouble?
2009 Mar 25
1
More on SIP for Skype
Daniel wrote:
For us, opensky can be OK for individual users, not for allowing
Asterisk users to call Skype users. Why? Simply that when you buy the 20
USD connection to Skype and don't want your calls to be cutted after 5
mn, you have to use the Gizmo Skype aliases system which is in your
account. Not really helpful if you want to connect transparently your
users to Skype! They better had to
2009 Mar 25
1
Skype TO SIP (Was SIP to Skype)
From: "Guillermo Salas M." <gsalas at manta.telconet.net>
> http://www.gizmo5.com/opensky Free calls are available up to 5
> minutes. If you need longer calls there's a commercial service you can
> purchase.
> Can be used to receive calls from skype?
Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it
will ring the IP phone connected to
2009 Feb 17
0
Questions about OpenSky - Asterisk to Skype Gateway
>> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
>>
>> > Hi there,
>> >
>> > is gizmo the first user of the Digium Skype solution, or do they use a
>> > different approach/product - any clue?
>> >
>> > http://www.gizmo5.com/pc/opensky/
>> >
>> > Philipp
OpenSky is no related to any product from Digium.
2009 Mar 27
0
SIP for Skype Solutions: Hosted v Non-hosted
2009/3/27 Marco Sambo <derwidtel at gmail.com>
> I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
> more invasive than Gizmo5 opensky. Doesn't it?
Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning
there's no software to install on your system. In minutes the system can be
working for your Asterisk box. This is like using