[Ekiga-list] Calling another voip provider
Hi all, How can I call a user of voip.com if I just have their username/sip name. Possible? Thanks. _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] quick help appreciated: ekiga always report symmetric NAT even after port-forwarding enabled.
version 2. I'm not having a problem that works for me. was posting the link for the other poster to see if it helped him.> Date: Sun, 28 Dec 2008 22:26:26 -0500> From: an...@bwh.harvard.edu> To: ekiga-list@gnome.org> Subject: Re: [Ekiga-list] quick help appreciated: ekiga always report symmetric NAT even after port-forwarding enabled.> > FS Inc. wrote:> > I am on a linksys router wrt54g that reports symentric NAT,> > > > What was recommended was to use port triggering as opposed to port > > forwarding. I am not too sure what the difference is, But basically i > > followed this link ...> > > > http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router (Symmetric > > NAT: Dynamic navigation of NAT routers section)> > This is exactly the way my father has his Motorola 2210-02-1006 router > configured. But it no longer works for him with Ekiga 3. Which version > are you using?> _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] quick help appreciated: ekiga always report symmetric NAT even after port-forwarding enabled.
I am on a linksys router wrt54g that reports symentric NAT,What was recommended was to use port triggering as opposed to port forwarding. I am not too sure what the difference is, But basically i followed this link ...http://wiki.ekiga.org/index.php/Ekiga_behind_a_NAT_router (Symmetric NAT: Dynamic navigation of NAT routers section)> Date: Mon, 29 Dec 2008 10:55:19 +0800> From: zhangwe...@realss.com> To: ekiga-list@gnome.org> Subject: [Ekiga-list] quick help appreciated: ekiga always report symmetric NAT even after port-forwarding enabled.> >1. If I set my notebook as DMZ, ekiga report I have cone NAT and> thus should work. But I cannot keep my notebook as DMZ for local> network management reasons;>2. If I don't set my notebook as DMZ, ekiga always report I have> symmetric NAT even if I enable port-forwarding strictly according> to recommendation;> 1. router (act as NAT firewall) setting is attached to this> email. Note 192.168.1.101 is IP address of my notebook;> 2. there is only 1 public IP address for this router;> 3. router setting is tested using netcat (running 'nc -u -l -p> 5062' on my notebook and run 'nc -u emerson.realss.com 5062'> on a remote server, test UDP port forwarding works fine by> typing hello world on remote server and see it comes out> locally);> > > Fast help highly appreciated because I am trying to start an online meeting.> > Other information: a) screenshot of ekiga is provided; b) netstat -l on> notebook attached as following> > zhangwe...@esmeralda:~$ netstat -ltu> Aktive Internetverbindungen (Nur Server)> Proto Recv-Q Send-Q Local Address Foreign Address> State> tcp0 0 *:sunrpc*:*> LISTEN> tcp0 0 *:x11 *:*> LISTEN> tcp0 0 *:http *:*> LISTEN> tcp0 0 *:ssh *:*> LISTEN> tcp 0 0 localhost:ipp *:*> LISTEN> tcp0 0 192.168.1.:h323hostcall *:*> LISTEN> udp0 0 192.168.1.255:5060 *:*> > udp0 0 192.168.1.101:5060 *:*> > udp0 0 *:bootpc*:*> > udp0 0 *:bootpc*:*> > udp0 0 *:5063 *:*> > udp0 0 *:5064 *:*> > udp0 0 *:5065 *:*> > udp 0 0 *:5066 *:*> > udp0 0 *:sunrpc *:*> > udp0 0 *:ipp *:*> > > -- > Real Softservice> > Huateng Tower, Unit 1788> Jia 302 3rd area of Jinsong, Chao Yang> > Tel: +86 (10) 8773 0650 ext 603> Mobile: 159 7382> http://www.realss.com> > _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ ekiga-list mailing list ekiga-list@gnome.org http://mail.gnome.org/mailman/listinfo/ekiga-list
Re: [Ekiga-list] Unable to get incoming calls from sip provider ekiga.net set up on asterisk
Are you sure? As I am using the ekiga wiki examples, the incoming works as
described but the outgoing is confusing to me as its not clear to follow the
example in the outgoing section of the ekiga wiki help.> From:
dsand...@seconix.com> To: ekiga-list@gnome.org> Date: Thu, 25 Dec 2008 11:53:02
+0100> Subject: Re: [Ekiga-list] Unable to get incoming calls from sip
providerekiga.net set up on asterisk> > I suppose you should ask this
question on the Asterisk mailing list ?> > Le mercredi 24 décembre 2008 à 21:50
+, FS Inc. a écrit :> > *bump*> > > > > >
__> > From:
fsh...@hotmail.com> > To: ekiga-list@gnome.org> > Date: Thu, 18 Dec 2008
00:39:46 +> > Subject: [Ekiga-list] Unable to get incoming calls from sip
provider> > ekiga.net set up on asterisk> > > > I've been using the wiki as a
guide here...> > > >
http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net> > > > I've
also tried various permutations, combinations and contortions but> > nada.> > >
> below is my extensions.conf> > > > * begin extensions.conf > >
[general]> > static = yes> > writeprotect = no> > autofallthrough = yes> >
clearglobalvars = no> > > > [globals]> > CONSOLE = Console/dsp ; Console
interface for demo> > ;CONSOLE=Zap/1> > ;CONSOLE=Phone/phone0> > IAXINFO =
guest ; IAXtel username/password> > ;IAXINFO=myuser:mypass> > TRUNK = Zap/G2
; Trunk interface> > TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0)> >
;TRUNK=IAX2/user:p...@provider> > > > [default]> > exten =>
s,1,Verbose(1|Unrouted call handler)> > exten => s,n,Answer()> > exten =>
s,n,Wait(1)> > exten => s,n,Playback(tt-weasels)> > exten => s,n,Hangup()> > >
> [macro-voicemail]> > exten => s,1,Dial(${ARG1},20)> > exten =>
s,n,Goto(s-${DIALSTATUS},1)> > exten =>
s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})> > exten => s-NOANSWER,n,Hangup()> >
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})> > exten => s-BUSY,n,Hangup()> >
exten => _s-.,1,Goto(s-NOANSWER,1)> > > > [incoming_calls]> > exten =>
ekiga_meyamma_in,1,Macro(voicemail,SIP/101)> > exten =>
ekiga_meyamma_in,n,Hangup()> > > > [internal_calls]> > exten =>
101,1,Macro(voicemail,SIP/101)> > exten => 101,n,Hangup()> > exten =>
102,1,Dial(SIP/102)> > exten => 102,n,Hangup()> > exten =>
8,1,VoiceMailMain(s${CALLERIDNUM})> > exten => 8,n,Hangup()> > exten =>
600,1,Answer()> > exten => 600,n,Playback(demo-echotest)> > exten =>
600,n,Echo()> > exten => 600,n,Playback(demo-echodone)> > exten =>
600,n,Hangup()> > > > [outgoing_calls]> > exten =>
_9.,1,Dial(SIP/ekiga_meyamma_out/${EXTEN:1},20,r))> > exten => _9.,n,Hangup()>
> > > [home]> > include => internal_calls> > include => outgoing_calls> > *
end extensions.conf > > > > below is my sip.conf> > > > begin sip.conf
***> > [general]> > context=default > > bindport=5060
> > > > bindaddr=0.0.0.0
> > srvlookup=yes > > > >
disallow=all > > allow=ulaw > >
allow=alaw> > allow=ilbc > > allow=gsm> > allow=h261> > > >
videosupport=yes> > > > ;Register 2345 at sip provider
'sip_proxy'. Calls from this> > provider> > ;connect to local extension
1234 in extensions.conf, default> > context,> > ;unless you configure a
[sip_proxy] section below, and configure a> > ;context.> > ;Tip 1:
Avoid assigning hostname to a sip.conf section like> > [provider.com]> > ;
Tip 2: Use separate type=peer and type=user sections for SIP> > providers> > ;
(instead of type=friend) if you have calls in both> > directions> >
register => meyamma:meya...@ekiga.net/ekiga_meyamma_in> > > >
externhost=fspublic.selfip.com> > externrefresh=10;> >
localnet=192.168.1.0/255.255.255.0> > > > [authentication]> > > > > > ;setup> >
[101]> > type=friend> > username=101> > secret=welcome> > qualify=yes> >
nat=no > >
Re: [Ekiga-list] Unable to get incoming calls from sip provider ekiga.net set up on asterisk
*bump*From: fsh...@hotmail.comto: ekiga-l...@gnome.orgdate: Thu, 18 Dec 2008
00:39:46 +Subject: [Ekiga-list] Unable to get incoming calls from sip
provider ekiga.net set up on asterisk
I've been using the wiki as a guide
here...http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.netI've
also tried various permutations, combinations and contortions but nada.below is
my extensions.conf* begin extensions.conf [general]static =
yeswriteprotect = noautofallthrough = yesclearglobalvars = no[globals]CONSOLE =
Console/dsp ; Console interface for
demo;CONSOLE=Zap/1;CONSOLE=Phone/phone0IAXINFO = guest ; IAXtel
username/password;IAXINFO=myuser:mypassTRUNK = Zap/G2 ; Trunk
interfaceTRUNKMSD = 1 ; MSD digits to strip (usually 1 or
0);TRUNK=IAX2/user:p...@provider[default]exten => s,1,Verbose(1|Unrouted call
handler)exten => s,n,Answer()exten => s,n,Wait(1)exten =>
s,n,Playback(tt-weasels)exten => s,n,Hangup()[macro-voicemail]exten =>
s,1,Dial(${ARG1},20)exten => s,n,Goto(s-${DIALSTATUS},1)exten =>
s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})exten => s-NOANSWER,n,Hangup()exten =>
s-BUSY,1,Voicemail(b${MACRO_EXTEN})exten => s-BUSY,n,Hangup()exten =>
_s-.,1,Goto(s-NOANSWER,1)[incoming_calls]exten =>
ekiga_meyamma_in,1,Macro(voicemail,SIP/101)exten =>
ekiga_meyamma_in,n,Hangup()[internal_calls]exten =>
101,1,Macro(voicemail,SIP/101)exten => 101,n,Hangup()exten =>
102,1,Dial(SIP/102)exten => 102,n,Hangup()exten =>
8,1,VoiceMailMain(s${CALLERIDNUM})exten => 8,n,Hangup()exten =>
600,1,Answer()exten => 600,n,Playback(demo-echotest)exten => 600,n,Echo()exten
=> 600,n,Playback(demo-echodone)exten => 600,n,Hangup()[outgoing_calls]exten =>
_9.,1,Dial(SIP/ekiga_meyamma_out/${EXTEN:1},20,r))exten =>
_9.,n,Hangup()[home]include => internal_callsinclude => outgoing_calls* end
extensions.conf below is my sip.conf begin sip.conf
***[general]context=default bindport=5060
bindaddr=0.0.0.0srvlookup=yes
disallow=all
allow=ulaw allow=alawallow=ilbc
allow=gsmallow=h261videosupport=yes;Register 2345 at sip
provider 'sip_proxy'. Calls from this provider;connect to local extension
1234 in extensions.conf, default context,;unless you configure a
[sip_proxy] section below, and configure a;context.;Tip 1: Avoid
assigning hostname to a sip.conf section like [provider.com];Tip 2: Use
separate type=peer and type=user sections for SIP providers; (instead
of type=friend) if you have calls in both directionsregister =>
meyamma:meya...@ekiga.net/ekiga_meyamma_inexternhost=fspublic.selfip.com
externrefresh=10
;localnet=192.168.1.0/255.255.255.0[authentication];setup[101]type=friendusername=101secret=welcomequalify=yes
nat=no host=dynamic canreinvite=no context=home;port=5061
;setup[102]type=friendusername=102secret=welcomequalify=yesnat=no
host=dynamic canreinvite=no context=home;port=5061
;ekiga.net[ekiga_meyamma_out]type=peerusername=meyammasecret=meyammahost=ekiga.netcanreinvite=noqualify=300insecure=port,invite;ekiga.net[ekiga_meyamma_in]type=userusername=meyammasecret=meyammahost=ekiga.netcanreinvite=noqualify=300context=incoming_callsinsecure=port,invite
end sip.conf These are the latest incantations of my sip and extensions
conf files. Can anyone assist in helping me to get incoming calls from
ekiga.net set up. I have the outgoing working as that was the simple part.
That one is handled by [ekiga_meyamma_out] and [outgoing_calls].Thanks a
milExplore the seven wonders of the world Learn more!
_
More than messages–check out the rest of the Windows Live™.
http://www.microsoft.com/windows/windowslive/___
ekiga-list mailing list
ekiga-list@gnome.org
http://mail.gnome.org/mailman/listinfo/ekiga-list
[Ekiga-list] Unable to get incoming calls from sip provider ekiga.net set up on asterisk
I've been using the wiki as a guide here...
http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net
I've also tried various permutations, combinations and contortions but nada.
below is my extensions.conf
* begin extensions.conf
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
[globals]
CONSOLE = Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO = guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK = Zap/G2 ; Trunk interface
TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:p...@provider
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,n,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)
[incoming_calls]
exten => ekiga_meyamma_in,1,Macro(voicemail,SIP/101)
exten => ekiga_meyamma_in,n,Hangup()
[internal_calls]
exten => 101,1,Macro(voicemail,SIP/101)
exten => 101,n,Hangup()
exten => 102,1,Dial(SIP/102)
exten => 102,n,Hangup()
exten => 8,1,VoiceMailMain(s${CALLERIDNUM})
exten => 8,n,Hangup()
exten => 600,1,Answer()
exten => 600,n,Playback(demo-echotest)
exten => 600,n,Echo()
exten => 600,n,Playback(demo-echodone)
exten => 600,n,Hangup()
[outgoing_calls]
exten => _9.,1,Dial(SIP/ekiga_meyamma_out/${EXTEN:1},20,r))
exten => _9.,n,Hangup()
[home]
include => internal_calls
include => outgoing_calls
* end extensions.conf
below is my sip.conf
begin sip.conf ***
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=h261
videosupport=yes
;Register 2345 at sip provider 'sip_proxy'. Calls from this provider
;connect to local extension 1234 in extensions.conf, default context,
;unless you configure a [sip_proxy] section below, and configure a
;context.
;Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
;Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions
register => meyamma:meya...@ekiga.net/ekiga_meyamma_in
externhost=fspublic.selfip.com
externrefresh=10;
localnet=192.168.1.0/255.255.255.0
[authentication]
;setup
[101]
type=friend
username=101
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=home
;port=5061
;setup
[102]
type=friend
username=102
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=home
;port=5061
;ekiga.net
[ekiga_meyamma_out]
type=peer
username=meyamma
secret=meyamma
host=ekiga.net
canreinvite=no
qualify=300
insecure=port,invite
;ekiga.net
[ekiga_meyamma_in]
type=user
username=meyamma
secret=meyamma
host=ekiga.net
canreinvite=no
qualify=300
context=incoming_calls
insecure=port,invite
end sip.conf
These are the latest incantations of my sip and extensions conf files. Can
anyone assist in helping me to get incoming calls from ekiga.net set up. I
have the outgoing working as that was the simple part. That one is handled by
[ekiga_meyamma_out] and [outgoing_calls].
Thanks a mil
_
Explore the seven wonders of the world
http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE___
ekiga-list mailing list
ekiga-list@gnome.org
http://mail.gnome.org/mailman/listinfo/ekiga-list
