Displaying 20 results from an estimated 500 matches similar to: "Asterisk as SIP UAC !!!"
2005 Mar 11
0
Errors using Asterisk as Sip Client behind SER !!!
Hi everyone,
I'm having some errors using Asterisk for incoming/outgoing calls. I have SER working with mysql without problems, all my internal users autenticate at SER and then if any number begins with a "1", Ser forwards the call to asterisk. Asterisk takes the forward and act as a Sip Client to make the call. I need to do that because My VoIP numbers are at Go2call, and I need
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
2005 Feb 02
0
SIP Call through Asterisk
I'm configuring my SER to forward calls based in extension. Cause I would like my ASTERISK to do international calls. How could I make ASterisk do international calls ?? I must pass the host (Go2Call), username and password to get the call up, but I don't know how.
I'm trying to find a extension command that like Dial, does the call but passing username, password and host for
2005 Feb 01
0
Asterisk Services working with SER !!!
Hi everybody,
I'm new to this list, my name is Felipe Martins and work for a telecom company. I'm interested in VoIP server to work as a service for my clients, I've already configured SER to work with mysql databases and authenticate all my users, two sip phones are already communicating with eash other, I can see all the logs generated by them, what I want now is to use asterisk as
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 Mar 14
2
asterisk outbound to SIP provider problems
Hi
I am having alot of difficulty connecting to SIP providers (I am trying 3)
and can't seem to find anything similar in the wiki or on the lists.....I
can receive inbound calls fine however on placing an outbound call the
calling phone never gets 'connected' but 2 way audio is passed for about
20secs before some sort of timeout.
Anything suggestions as to what I could try
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 Dec 12
1
AGI problema
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Verdana">Hi all. I've written a AGI in C language.
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
2010 Dec 12
2
CMCI exceptions happened and MCE entry state transition made Xen crashed.
Hi all,
Three days ago, the server reported lots of CMCI exceptions and Xen 3.4.2 printed hundreds of "CMCI: send CMCI to DOM0 through virq" messages to the console . From the console output, Then I can see that Dom0 try to read the MSR_CAP regs by #GP trap in order to log the MCA error.
I am not sure why so many CMCI happened , maybe there were some thing wrong with the hardware.
2010 Dec 12
2
CMCI exceptions happened and MCE entry state transition made Xen crashed.
Hi all,
Three days ago, the server reported lots of CMCI exceptions and Xen 3.4.2 printed hundreds of "CMCI: send CMCI to DOM0 through virq" messages to the console . From the console output, Then I can see that Dom0 try to read the MSR_CAP regs by #GP trap in order to log the MCA error.
I am not sure why so many CMCI happened , maybe there were some thing wrong with the hardware.
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
2003 Jun 06
1
more about SIP ...
I added the line "allow G723.1" in my sip.conf general config,
and from a bridge connection which gives silence,
I have progressed to the error message below,
and the call gets rejected.
help!!
Dave
ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant
Expressa
723@216.52.153.207 : Go2Call SIP gateway
-- Executing
2007 Aug 03
0
Several doubts on Asterisk as an UAC
Hi,
I'm new to Asterisk and I've been trying to configure it to talk to
several SIP providers (such as FWD). I found that, although there are
some "recipes" on how to do it, there are few documents that really
explain *why* the settings are used, and overall I found very little
documentation on sip.conf.
I've been using this page as a reference:
2005 Dec 27
2
postfix, dovecot, sasl deliver error
Hi,
I am receiving an error when trying to send mail. I am using FreeBSD 6
and dovecot 1.0.alpha5 and postfix 2.3-20051223 which includes the
dovecot sasl patch.
I am getting:
Dec 26 17:26:45 example postfix/pipe[612]: DC90D5C30:
to=<tep@example.com>, relay=dovecot, delay=14, delays=14/0.05/0/0.08,
dsn=4.3.0, status=deferred (temporary failure. Command output: Error:
2004 Jul 19
0
Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
Hi, does anyone have the setup for go2call ?
I have digium boards and quicknet linejacks and phonejacks.
The cards work fine in asterisk without the g729 or g723.1 for the
phonejack.
I will like to do SIP origination using the codec in the phonejack and
linejack g729 or g723 and send the calls to go2call.
Anyone has the setup for this ? Or similar setup to a SIP provider using
g729 or g723
2003 May 13
1
beginner's question!
hi there,
I have just downloaded and installed asterisk a couple of days ago, it compiled correctly and starts up and runs, on a Redhat 9 system freshly installed for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips :
firstly on modifying the
2012 Jan 14
1
Asterisk as UAC: How to put call OnHold
Hi!
Maybe I am missing something or am a little blind at the moment, but I
didn't find out how asterisk can place a call on hold when acting as user
agent client to another SIP server.
Scenario:
----------
Asterisk registers to another SIP server (provider) as user agent.
An inbound call from this other SIP server comes in and arrives at asterisk.
Asterisk performs some actions in the
2007 Apr 12
4
Re: [Xense-devel] [RFC][PATCH][UPDATED] Intel(R) LaGrande Technology support
Hello,
Has any more work been done on this front? The message below is
from Sept. 2006. In particular, the LT/TXT Technology Enabling
Platform (TEP) is now available from MPC Corp. Where can one
obtain an appropriate AC SINIT module (i.e., like
lpg_sinit_20050831_pae.auth.bin below)? I would like to begin
using Xen with TXT support.
Thanks,
-Jon
This patch adds SMP support to the