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don Paolo Benvenuto

2006-Jun-12 19:49 UTC

[Asterisk-Users] transferring calls from ekiga to asterisk

I have ekiga registering to a voip provider (skypho) and receiving
external call
through the stun server.

I want to redirect inconditionally all these calls to my asterisk
server, but I can't understand how and what should I configure in
asterisk in order to accept the redirected call.

In asterisk console I can't see nothing when ekiga passes the call.

If I turn asterisk's sip debug, I can see that the call arrives to
asterisk from 0108392222@voip.eutelia.it (skypho provider) via something
containing my external IP address, and asterisk tries to communicate
with a host on my external IP address, obviously unsuccessfully, and in
ekiga I get a occupied tone.

Note that in ekiga I have an account which is in sip.conf, and ekiga
registers without problems with that account to my asterisk server.

However, the problem I have is how to transfer to asterisk a call which
is managed with another account, specifically a external voip provider
account: the call arrives to asterisk with the data of that external
voip provider.

Anyone could help me? Thank you!

-- 
Buon Cammino!

don Paolo Benvenuto

Vuoi sapere di pi? su quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario

Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/

don Paolo Benvenuto

2006-Jun-13 10:48 UTC

head link

[Asterisk-Users] transferring calls from ekiga to asterisk

El mar, 13-06-2006 a las 07:33 +0000, undrhil.1528785@bloglines.com
escribi?:> When you configured the incoming line in sip.conf, you gave it a context.
I think my problem is: how do I configure sip.conf in order to receive
those call redirects?

In Twinklephone I have two accounts:
- an account in which twinkle is a peer of asterisk, which has this
settings in sip.conf:

[pablopctwinkle]
type=friend
secret=xxxxxx
callerid="Pablo PC Twinklephone" <619>
host=dynamic
context=todo
nat=no
qualify=yes

twinkle registers with asterisk without problems with these settings. It
sends and receives calls, it's a normal asterisk's extension

- another with the voip provider (voip.eutelia.it)

This account is the one that receives the calls and redirects it to
asterisk.

I want to transfer a call from this account to asterisk. That's
equivalent, I think, to connecting to asterisk from that account.

196.3.84.214 my routers external address
5062 is the port that twinklephone uses
10.152.58.1=misiongenovesa is the server asterisk and twinklephone are
running on
0108937227 is my username with voip provider voip.eutelia.it

I must configure sip.conf and extensions.conf in order to receive calls
from that account.

If I try to call asterisk from that account I get in asterisk's console
(sip debug):

-----------BEGIN---------
<-- SIP read from 10.152.58.1:5062:
INVITE sip:600@misiongenovesa SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: <sip:600@misiongenovesa>
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
Contact: <sip:0108937227@196.3.84.214:5062>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (13 headers 14 lines)---
Using INVITE request as basis request - ouzpyxycwrkeptj@196.3.84.214
Sending to 196.3.84.214 : 5062 (NAT)
Found peer 'pablopctwinkle'
Reliably Transmitting (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Scheduling destruction of call 'ouzpyxycwrkeptj@196.3.84.214' in 15000
ms
Retransmitting #1 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Retransmitting #2 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Retransmitting #3 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Retransmitting #4 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Retransmitting #5 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0


---
Retransmitting #6 (no NAT) to 196.3.84.214:5062:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
To: <sip:600@misiongenovesa>;tag=as111cfbda
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:600@10.152.58.1>
Proxy-Authenticate: Digest realm="asterisk",
nonce="78099a8f"
Content-Length: 0

<-- SIP read from 10.152.58.1:5062:
INVITE sip:600@misiongenovesa SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: <sip:600@misiongenovesa>
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
Contact: <sip:0108937227@196.3.84.214:5062>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (13 headers 14 lines)---
Ignoring this INVITE request
<-- SIP read from 10.152.58.1:5062:
INVITE sip:600@misiongenovesa SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: <sip:600@misiongenovesa>
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
Contact: <sip:0108937227@196.3.84.214:5062>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


---
misiongenovesa*CLI>
<-- SIP read from 10.152.58.1:5062:
INVITE sip:600@misiongenovesa SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: <sip:600@misiongenovesa>
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
Contact: <sip:0108937227@196.3.84.214:5062>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

---
Destroying call '652b07e372f8380e52e59a7923d1d0ec@10.152.58.1'
12 headers, 0 lines sip no debug
Reliably Transmitting (NAT) to 195.62.225.244:5060:
OPTIONS sip:voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5060;branch=z9hG4bK5f6a5c22;rport
From: "asterisk" <sip:asterisk@196.3.84.214>;tag=as56330376
To: <sip:voip.eutelia.it>
Contact: <sip:asterisk@196.3.84.214>
Call-ID: 01dd513b49661b6b6887268709221f78@196.3.84.214
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 13 Jun 2006 17:40:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
misiongenovesa*CLI> sip no debug
<-- SIP read from 10.152.58.1:5062:
INVITE sip:600@misiongenovesa SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm
Max-Forwards: 70
To: <sip:600@misiongenovesa>
From: "don Paolo Benvenuto"
<sip:0108937227@voip.eutelia.it>;tag=cfpll
Call-ID: ouzpyxycwrkeptj@196.3.84.214
CSeq: 979 INVITE
Contact: <sip:0108937227@196.3.84.214:5062>
Content-Type: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY,
SUBSCRIBE
Supported: 100rel
User-Agent: Twinkle/0.7.1
Content-Length: 311

v=0
o=0108937227 1395986944 491937694 IN IP4 196.3.84.214
s=-
c=IN IP4 196.3.84.214
t=0 0
m=audio 8000 RTP/AVP 3 98 97 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

--- (13 headers 14 lines)---
Ignoring this INVITE request
Retransmitting #1 (NAT) to 195.62.225.244:5060:
OPTIONS sip:voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 196.3.84.214:5060;branch=z9hG4bK5f6a5c22;rport
From: "asterisk" <sip:asterisk@196.3.84.214>;tag=as56330376
To: <sip:voip.eutelia.it>
Contact: <sip:asterisk@196.3.84.214>
Call-ID: 01dd513b49661b6b6887268709221f78@196.3.84.214
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 13 Jun 2006 17:40:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


------------END----------





>  This context needs to be defined in extensions.conf.
> 
> For example, if you
> defined your incoming SIP line with context=incoming_sip, then in
extensions.conf
> you would define:
> 
> [incoming_sip]
> exten => s,1,Answer
> exten => s,2,VoicemailMain
> 
> exten => s,3,Hangup
> 
> Something along those lines to tell Asterisk what
> you want that incoming call to do.
> 
> If this doesn't help, then maybe if
> you post your sip.conf and extensions.conf and a capture of the CLI when a
> call is incoming, someone might be able to help you out.  :)
> 
> Undrhil
> 
> 
> --- donpaolo@gsi.it wrote:
> I have ekiga registering to a voip provider (skypho)
> and receiving
> > external call
> > through the stun server.
> > 
> > I want to
> redirect inconditionally all these calls to my asterisk
> > server, but I can't
> understand how and what should I configure in
> > asterisk in order to accept
> the redirected call.
> > 
> > In asterisk console I can't see nothing when ekiga
> passes the call.
> > 
> > If I turn asterisk's sip debug, I can see that the
> call arrives to
> > asterisk from 0108392222@voip.eutelia.it (skypho provider)
> via something
> > containing my external IP address, and asterisk tries to
> communicate
> > with a host on my external IP address, obviously unsuccessfully,
> and in
> > ekiga I get a occupied tone.
> > 
> > Note that in ekiga I have an
> account which is in sip.conf, and ekiga
> > registers without problems with
> that account to my asterisk server.
> > 
> > However, the problem I have is
> how to transfer to asterisk a call which
> > is managed with another account,
> specifically a external voip provider
> > account: the call arrives to asterisk
> with the data of that external
> > voip provider.
> > 
> > Anyone could help
> me? Thank you!
> > 
> > -- 
> > Buon Cammino!
> > 
> > don Paolo Benvenuto
> > 
> 
> > Vuoi sapere di pi? su quello che succede qui?
> > leggi il mio diario a
> http://www.chiesamissionaria.it/diario
> > 
> > Visita l'enciclopedia libera,
> dove puoi contribuire anche tu:
> > http://it.wikipedia.org/
> > 
> > _______________________________________________
> 
> > --Bandwidth and Colocation provided by Easynews.com --
> > 
> > Asterisk-Users
> mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> > -- 
Buon Cammino!

don Paolo Benvenuto

Vuoi sapere di pi? su quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario

Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/

don Paolo Benvenuto

2006-Jun-27 18:08 UTC

head link

[Asterisk-Users] transferring calls from ekiga to asterisk

I have ekiga registering to a voip provider (skypho) and receiving
        external call
        through the stun server.
        
        I want to redirect inconditionally all these calls to my
        asterisk
        server, but I can't understand how and what should I configure
        in
        asterisk in order to accept the redirected call.
        
        In asterisk console I can't see nothing when ekiga passes the
        call.
        
        If I turn asterisk's sip debug, I can see that the call arrives
        to
        asterisk from 0108392222@voip.eutelia.it (skypho provider) via
        something
        containing my external IP address, and asterisk tries to
        communicate
        with a host on my external IP address, obviously unsuccessfully,
        and in
        ekiga I get a occupied tone.
        
        Note that in ekiga I have an account which is in sip.conf, and
        ekiga
        registers without problems with that account to my asterisk
        server.
        
        However, the problem I have is how to transfer to asterisk a
        call which
        is managed with another account, specifically a external voip
        provider
        account: the call arrives to asterisk with the data of that
        external
        voip provider.
        
        Anyone could help me? Thank you!
        
-- 
Buon Cammino!

don Paolo Benvenuto

Vuoi sapere di pi? su quello che succede qui?
leggi il mio diario a http://www.chiesamissionaria.it/diario

Visita l'enciclopedia libera, dove puoi contribuire anche tu:
http://it.wikipedia.org/

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