Displaying 20 results from an estimated 2000 matches similar to: "How to put peers into Realtime"
2006 Oct 24
0
sip.conf - srvlookup
I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get
voipbuster/tomo 194.221.62.207 5060 OK (27 ms)
And when I ping sip1.voipbuster.com
[root@tomo ~]# ping sip1.voipbuster.com
PING sip1.voipbuster.com
2006 Mar 28
0
codec translation problem???
2006 Jan 27
2
Name/username (sip show peers)
How can I make it more readable?
Name/username
601/601
123456789/123456789
voipbuster/abcd
601 = hotline
123456789 = Peter Pan
only voipbuster/abcd is easy read/understandable!
bye
Ronald Wiplinger
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
2006 Jan 21
1
Is sip1.voipbuster.com corking reliably for others on list?
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling out, I see that there is SIP exchange, and in many
cases also RTP data being exchanged.
Hover in a very large number of attempts the connection is not
established. Half of the time there is no RTP, the rest of the time there
*is* RTP data flowing in two ways, but no ringtone is
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 23
1
How to set-up LCR
How to set-up LCR ?
a. which companies can be used with LCR?
b. how to set-up & maintain LCR?
c. multiple connection to one gateway?
Example:
+886223456789 could be reachable via
a. ENUM free
b. Dundi free
c. Voipstunt free
d. Voipbuster free
e. Nufone $
f. Voipstunt $
g. others with 4 concurrent connections $$
h. others with 3 concurrent connections $$
I am looking
2008 Apr 16
2
about getting webservice from other website
I have a webservice offered by a website to be integrated for my
website.They ask me to follow three steps to finish integrating.These
steps are:
1. Create an HTTP connection.
2. Send request parameters via POST method.
3. Parse XML-formatted response string.
The request parameters are something like following:
"Version=2.0.0.0" +
"&ShipmentID=1234" +
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
2005 Jul 17
2
DNS SRV
I have added in my zone file;
_sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com.
As I understand it should mean that any sip connection to
<anyname>@elmit.com should go to the udp port 5060 at the host
vpb.elmit.com.
In Asterisk's extensions.conf I have in the context [default]
exten => ronald,1,Dial(${PHONE_615},60,tr)
exten => ronald,2,Voicemail,u615@office
exten =>
2006 Apr 02
1
Who is on a call?
I would like to know which extension number is engaged in a call.
show channels shows me:
*CLI> show channels
Channel Location State
Application(Data)
SIP/asterisk.elmit.com-0 690@default:2 Up
Echo()
SIP/8807-066 690@newcontext Up Echo()
2 active channels
2 active calls
but it is not
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 May 17
4
Is SKYPE a threat or should we do something (together)
Skype is very succesfsfull and get more and more users, ... we can
ignore them, accept them or do something,...
My suggestion is that we try to do something, ...
If we would peer to each other, than we get soon also a great amount of
users together, and than our service becomes more valuable, ...
Let's discuss advantages and disadvantages!
bye
Ronald
--
Ronald Wiplinger (CEO of
2007 Feb 14
5
Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the
bandwidth to different users. Each user should get e.g., 512kB/s plus
256kB/s dedicated for VoIP.
What kind of device can I use for that ? (managing switch ??? which one?)
bye
Ronald Wiplinger
2006 Jun 14
0
Strange problem with MusicOnHold - works outgoing - works with extension - but not incoming!
I've got a strange situation with musiconhold.
It works if I dial my extension 6000:
>From extensions.conf:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()
Debug output if I call 6000:
-- Executing Answer("SIP/gs1-b6ee", "") in new stack
-- Executing MusicOnHold("SIP/gs1-b6ee", "") in new stack
-- Started music on hold,
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a
link on a web site to a webphone on MY SITE !!!
Has anybody an idea for that? AJAX?
bye
Ronald Wiplinger
2006 Oct 17
0
lots of registrations, sip problem
Hello,
I've got a problem with connection to my SIP provider. In general,
everything works, but I get lots of these messages:
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's
odd... Got a response on a call we dont know about. Cseq 42710 Cmd
SIP/2.0
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request:
That's odd... Got a response on a call
2005 Jul 28
12
Can you caculate with me?
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers: 0.7269 or 0.2929 ???
bye
Ronald Wiplinger