Displaying 20 results from an estimated 500 matches similar to: "more about SIP ..."
2006 Dec 12
1
AGI problema
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<font face="Verdana">Hi all. I've written a AGI in C language.
2003 May 29
1
a beginner's SIP question ..
I am trying to get asterisk to dial this address :
sip:723@216.52.153.207
Using a softphone on my PC (217.168.168.49)
it dials immediately and I get a voice prompt ..
I have configured an extension, 1303 on asterisk,
modifying the demo configuration :
exten => 1303,1,Dial(SIP/723@216.52.153.207)
When from my softphone I dial
sip:1303@217.168.168.51
on the console I get :
-- Executing
2006 Nov 02
1
is IAX required for firewall and router?
I'm trying to understand IAX and whether or not it would solve my
difficulties:
'The primary goals for IAX were to minimize bandwidth used in media
transmissions, with particular attention drawn to control and individual
voice calls, and to provide native support for NAT (Network Address
Translation) transparency. Another goal is to be easy to use behind
firewalls.'
2003 Jun 06
3
SIP codecs
i've been having a problem getting two SIP phones
to bridge running through asterisk, actually one is
a SIP softphone, SJ Phone, and the other is the
Go2Call calling gateway.
Someone suggested that I don't have the right codecs.
How do I find out which codecs are installed, and how
can I install further codecs? Any suggestions which
would be the right one?
I think hte problem is from the
2003 May 27
1
please help (reposted) - re. * connecting to a commercial call service
hi,
maybe someone out there already has some experience and can help me.
I have just ordered an E100P card from Digium, I already have a basic
asterisk setup up & running.
My application is the following :
I want to accept incoming calls from the PSTN to Asterisk, and without
asking anything of the client just pass them immediately to a call gateway
in USA, actually we are planning to use
2006 Jul 17
2
Quantreg error
Dear User,
I got the following error running a regression quantile:
> rq1<-rq(dep ~ ., model=TRUE, data=exo, tau=0.5 );
> summary(rq1)
Erro em rq.fit.fnb(x, y, tau = tau + h) :
Error info = 75 in stepy: singular design
Any hint about the problem?
Thanks a lot,
________________________________________
Ricardo Gon?alves Silva, M. Sc.
Apoio aos Processos de Modelagem Matem?tica
2006 Jul 26
3
Moving Average
Dear R-Users,
How can I compute simple moving averages from a time series in R?
Note that I do not want to estimate a MA model, just compute the MA's
given a lenght (as excel does).
Thanks
________________________________________
Ricardo Gonçalves Silva, M. Sc.
Apoio aos Processos de Modelagem Matemática
Econometria & Inadimplência
Serasa S.A.
(11) - 6847-8889
ricardosilva@serasa.com.br
2004 Jan 09
2
asterisk sip with voicemail
Hello all,
I have setup my sip.conf so users can register etc in the following
format,
[person]
type=friend
username=nick
secret=********
host=dynamic
mailbox=101
in my voicemail.conf I have an entry like
101 => 1234,Nick Knight,nick@omniis.com
Leaving a voicemail works fine after I have my dial command time out but
on sip clients which display whether voicemail is
2006 Jul 13
2
MLE and QR classes
Hi,
I load my data set and separate it as folowing:
presu <- read.table("C:/_Ricardo/Paty/qtdata_f.txt", header=TRUE, sep="\t",
na.strings="NA", dec=".", strip.white=TRUE)
dep<-presu[,3];
exo<-presu[,4:92];
Now, I want to use it using the wls and quantreg packages. How I change the
data classes for mle and rq objects?
Thanks a lot,
2004 Jan 20
0
Outbound call with Go2Call
Any got experience with these?
I couldn't fint anything in any postings...
it seems they have a h.323 on voip01.go2call.com and a sip on
sip01.go2call.com
I have tried to register with some of the same as I use for nikotel, but
Asterisk does not want to register.
I've tried to use both the user name (ingvald) and the PIN code 440.... as
authentication.
---from sip.conf----
2006 Apr 28
1
Warning: No path to translate with SJPhone
Hi list!
I'm making tests for Asterisk. I've tested with 2 users installing SJphone
and it worked fine, but when I install it over a third user with the
softphone, the phone dial for 2 seconds and a window alert goes out on the
softphone:
Busy
Call rejected: 486 Busy Here
And on my Asterisk server this message:
Apr 28 09:05:37 WARNING[8140]: channel.c:2685 ast_channel_make_compatible:
2006 Jun 14
1
transcoding problem
I am having a problem with asterisk transcoding GSM and G729 codecs, the
error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible:
No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2)
Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop
call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3
==
2005 May 05
2
Did nufone change allowed codecs?
Hi,
I've been using nufone DIDs for months with no problem. Now there are
codec problems that prevent any kind of calls working. For example,
May 5 13:04:12 WARNING[928]: channel.c:2115
ast_channel_make_compatible: No path to translate from
IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4)
May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop
call because I couldn't
2005 Feb 02
0
Problemas with Basic Services.
Hi Everybody,
I'm trying to make my asterisk dial a international call from a SER request of it. My ambient is like this.
[Clients]--[SER]--[Asterisk]--[Go2Call]
Client: My SIP clients.
SER: My REGISTRAR/Proxy Server
Asterisk: All other services(Voicemail,musiconhold etc) and also acting as an UAC dialing International Calls, because SER doesn't do that sending username, password and
2010 Nov 12
1
Call failed becaus of SIP tanslate
Hi Guys,
I have a the following issue when I ma trying to place a call through my
voip provider, I am using an asterisk 1.2.21.1, do you have an idea what
could fix this issue (as you can see when the other party answered, the call
get dropped because of probably sip incompatibility)
Nov 12 14:31:30 WARNING[21432]: chan_sip.c:2587 sip_write: Asked to transmit
frame type 256, while native formats
2005 Jul 11
2
Unable to dial certain calls
To begin with, I am new to both asterisk and VOIP and although I've
gotten pretty far with my Asterisk setup and have two different sip
accounts working fine for outgoing calls I am having trouble with one
issue.
My problem is that I have another provider who uses IAX2 that I wish
to use for calling various countries, including local (The
Netherlands) calls and calls to the US to
2006 Dec 15
2
call from h323 to SIP
Hi
i am trying to do the same thing:
receive a call from a cisco callmanager and forward it to a SIP user.
Asterisk is compiled with h323 support, and is configured as a gateway
in the cisco callmanager.
h323.conf:
[general]
port = 1720
bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP
address for this machine
allow=all
extension.conf:
exten = 3298,1,Answer
exten =
2005 Jun 17
1
Unable to find a path from g729 to gsm
Greetings! to all
Now, with some hard time and help from many genurous people's in the list, I
have come to this point with my TDM20B card & my teliax's IAX2 account.
I hope someone may help me with this issue mentioned below. I have already
selected my codec as gms in my iax.conf as well as in teliax's "my account
page" but still i have the same error when I attempt
2005 Feb 11
0
Asterisk as a UAC forwarded by SER
Hi everybody,
I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid
2005 Feb 14
0
Asterisk as SIP UAC !!!
Hi gentleman
I've configured SER to forward every call starting with sip uri request "1" to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it call to my other SIP Provider outside my network, sending username and password for authentication.
I've read at www.voip-info.org some articles but found none that could suit to my needs, but yet I've found an