Displaying 20 results from an estimated 100 matches similar to: "Outbound call with Go2Call"
2004 Jan 16
7
CAPI not installed, after changed from i4l to CAPI
I had unexpected hangups from my asterix box using the i4l driver. (SIP
<-> SIP calls worked execellent, but SIP<->ISDN didn't.)
Then I changed the i4l driver in modem.conf with the chan_capi from
jungham. (http://www.junghanns.net/asterisk)
I followed his instructions in the INSTALL file, and first encountered
some errors compiling it. It help by deinstalling several
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 Jul 10
2
OH323 + G729 + Go2Call
hi ..
i've just installed and licensed an instance of the G729 codec.
I am trying to connect through asterisk to Go2Call server ..
According to their info it involves dialling extension 729 on
voip01.go2call.com, to get the IVR.
my extensions.conf shows :
exten => s,2,Dial(OH323/h323:729@216.52.153.206)
which I think is correct, I have G729 enabled in the OH323.conf
file and it seems to
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
2006 Dec 12
1
AGI problema
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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 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
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
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
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
2004 Aug 23
1
using ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular extension is
available in the sip channel
I am using MySQL to hold my SIP friends.
and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip channel, I
will send the call somewhere else
my extensions file contains the following:
exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten
2010 Jan 04
0
Domain Trusts between 3.0.28 and 3.4.3 with '--enable-merged-build'
Hi,
Is is possible to have inter domain trusts between Samba 3.0.28 and
Samba 3.4.3, the latter compiled with the '--enable-merged-build'
option?
(I.e. similar to
<URL:http://www.samba.org/samba/docs/man/Samba-HOWTO-Collection/InterdomainTrusts.html#id2621030>
but with Samba 3.4.3 instead of Windows 2000.)
--
Best regards,
Knut Ingvald Dietzel
-------------- next part
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
2002 Feb 26
1
Cross-tabulation of data from database
I am quite new to R, so please bear over with me if I have problems with the
R terminology.
I want to (try to) use R for some analyses within vegetation ecology, using
the vegan package. I have my data in a postgresql database, and I manage to
get them into R as a dataframe with columns for respectively: Name of the
analysed m2, Name of the species, coverage of species in the square in %. I
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
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
2004 Nov 27
0
Failed to WWW-authenticate on INVITE
I'm having trouble connecting a asterisk server to a SIP Express router.
Inbound calls to my asterisk server works just fine, but when i try to
make outbound calls I get the following error message:
Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to
WWW-authenticate on INVITE to '"username"
<sip:username@mysipprovider>;tag=as5399a078'
I'm
2004 Nov 30
5
cisco dial-peer voip
I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over
pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y
So far so good.
But I want to setup VOIP sessions with local carrier. I added dial-peer
40 for this. Session target x.x.x.x But calls will always get routed to
the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried.
My situation:
PSTN
2004 Jan 07
1
Unexpected ISDN hangup on outbound call
We have setup an asterisk box to let everybody call into the university
internal network, but I get unexpected hangups when doing an outbound call
from SIP to the ISDN interface, and it happens from 20 seconds to some minutes into the
call.
----------the dial and the problem-----------
-- Executing Dial("SIP/57966-a19d", "Modem/g1:96121||rt|") in new
stack
--