Displaying 20 results from an estimated 1000 matches similar to: "Is sip1.voipbuster.com corking reliably for others on list?"
2007 Mar 24
2
freepbx -> DB Error messages...
Hi all,
I am probably missing something ultimately obvious, but I have a problem
configuring freepbx...
Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu
installation guide on freepbx.org.
System pxe-boots from a server with NFS root on same
Using * 1.2 current (from source, not .deb's)
Using mISDN-streams (from source, not .deb's)
Using freePBX-2.2.1 (from source, not
2006 Jan 20
1
AIX calls with sipdiscount
Hi
Someone have luck using Sipdiscount service with IAX ?
I only can use sipdiscount IAX service using a free account (only 1 minute
call) , I have a normal account and with it can login in the IAX server.
I using sip1.sipdiscount.com like IAX server but can make free calls (less 1
minute).
Thanks in advance.
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Servidores BSD, Solaris y Linux
2007 May 10
2
The downside of Asterisk and least cost routing...
I forgot to pay this month's phone bill, and never noticed until family
(the in-laws, who are too cheap to try the cell phone if landline fails,
because it is 'more expensive') told me they were unable to reach us...
As it turns out, the phone company disconnected us, but because Asterisk
routes all outgoing calls in the Netherlands over VoipBuster, I never
noticed anything! ;-)
If
2006 Jan 10
3
IAX & CallerID
Hi All
Apologises if this has been disussed and I missed it.
My SetUp
I have a sip phone registered to an asterisk box (a1) in one location 1.
This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone.
My Problem
The caller ID setup in the sip.conf for the phone registered to a1 is not
2006 Mar 24
3
Call terminated after 60 seconds
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
>From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no other
configuration than all their other users.
What can I do.
I removed all asterisk functionality by forwarding the inboud call
directly to a local
2006 Jan 22
3
Installing the none commercial intel g729codecs into Asterisk@Home 2.2?
Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it?
Thanks,
Doug.
-----Original Message-----
From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com]
Sent: Sun 1/22/2006 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
2006 Jan 23
1
Installing the none commercial intel g729 codecs into Asterisk@Home 2.2?
Yep I did the same.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Francesco Peeters (Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: fbraeuer@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
2009 Sep 02
1
Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
2006 Jan 06
3
Recording Calls at the phone
I work for a call center and we are looking at using asterisk to have
our operators take calls. Our message taking software records all the
calls on the operators computers. Right now we use these recording
controls from radio shack that plug in between the wall jack and the
phone and plug in via a 1/8 inch stereo connector to the mic input on
the computer. If I buy an IP phone I can't do
2006 Jan 27
1
Installing the none commercial intel g729 codecsinto Asterisk@Home 2.2?
Thanks but this is for a test, I didn't buy the first one as it's a non commercial installation. I'm trying to test bandwidth etc so I need to try out how 4 of them handle the link simultaneously, I just don't know how to add a second test license.
Dean
________________________________
From: asterisk-users-bounces@lists.digium.com
2006 May 29
4
registration at Voipbuster times out
Hi,
I am new here on this list, and have a problem of which I hope that somebody here can help me with it.
I have a Voipbuster account, with which I would like to make phone calls via my Asterisk PBX. If I let X-Lite register directly at voipbuster.com, everything is OK, but if I let Asterisk register there, it says "registration for XXXXXX@sip.voipbuster.com timed out, trying again",
2005 Sep 10
2
VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 11112
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is
2005 Sep 26
1
voipbuster advise
Hi,
I'm using voipbuster at work, and I've got 2 questions:
1) Is it possible to send faxes using voipbuster connex?
2) Is it possible to cut off or cover the voice that say the charge
per minute?(I've payed the '5' euro, and from that moment I've got
it!).
Of course I understand that is to let me know how much I'm going to
spend, but I do not like it, expecially when
2006 Jan 27
5
External IAX2 phone defined as internal behaving as from PSTN
Have asterisk@home 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring
2006 Jan 19
0
Incoming fax on voipbuster
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
problems.
For testing I've add a local number (300) to the dialplan. When I call
this number
2006 Jun 08
0
"I can hear them but they can't hear me" with VoipBuster
Hi;?
When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly.
I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example.
?
I tried with different codecs: gsm, alaw and ulaw but no change.
?
So, now?I
2005 Sep 19
1
Voipbuster in Australia -- delay problem
Hi, all,
I got my * to work with voipbuster service. And it works quite well when I
am calling USA or Europe. However, for local calls, I am experiencing long
delays (About 1s). As far as I know, voipbuster application does not have
this problem.
I am using IAX and gsm codec.
Any ideas on how to combat this?
Thanks,
Rudolf
2005 Aug 31
7
VoipBuster with astersisk?
Hi, all
Here is a something I found on the web:
http://www.voipbuster.com
And it works OK too. Now, I want to use it via asterisk, so I ccan use my normal phones instead of PC application.
Did anyone try to connect astersisk and VoipBuster?
Thanks,
Rudolf
2006 Jan 11
1
Zaptel modules load, but Asterisk fails at s tartup
/etc/conf.d/local.start
-----Original Message-----
From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com]
Sent: Wednesday, January 11, 2006 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zaptel modules load, but Asterisk fails at
startup
On Wed, January 11, 2006 21:36,
2006 Feb 05
1
1 ISDN BRI to IAX2/SIP... (*) best tool or?...
I have a question,
I have to provide a solution for an office that will be almost abandoned,
and there will be one or sometimes two persons 2 days a week. The main
number however should be preserved.
They have several ISDN BRI connections, most of which will be dropped.
Only one will be retained, for 2 reasons:
1) It has the ADSL link
2) The number has been the main contact number for over 20