[Asterisk-Users] Starting From Scratch

Wed Sep 14 21:34:11 MST 2005

Try this just dial 0 to dial out and then dial a internal number at you
company
What are the xxxx between 916 and 2128 ?
exten => _0.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},70,Tt)

-----Oorspronkelijk bericht-----
Van: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Namens Thczv F. Thczv
Verzonden: donderdag 15 september 2005 5:55
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Starting From Scratch

Hello all:

For fun, I am learning about Asterisk, and trying to get Asterisk working at
my house.  I installed Asterisk at Home.  It seems to be functioning fine.  I
installed a couple of softphones, and have them registered with Asterisk.  I
actually work for a CLEC, and I have registered my Asterisk box with SER
(which I don't begin to understand
yet) at the office.  In order to try to understand how all this works, I
have stripped my extensions.conf down to almost nothing.  I am building it
up piece by piece.  This is the entirety of my extensions.conf file:

[globals]
OUTBOUNDTRUNK=SIP/mysipprovider.com

[from-internal]
exten => 105,1,Answer()
exten => 105,2,Playback(abandon-all-hope) exten => 105,3,Hangup() exten =>
106,1,Dial(${OUTBOUNDTRUNK}/916xxx6000)
exten => 107,1,Dial(${OUTBOUNDTRUNK}/916xxx2128)

This is all just testing.  When I dial 105 from either of my softphones, it
plays the recording fine.  My thought for the 106 and
107 extensions was to sort of hard code it so that if I dialed either of
those extensions the call would automatically get routed over my outbound
sip trunk to the appropriate offsite PSTN dialable number. 
Once I know that the calls will go through, I will create a proper dial
plan.

The 6000 number is my home PSTN phone.  The 2128 number is my office desk
phone (a SIP phone).  Here is where I get stumped:  When I dial
107 from my SIP softphone, the call goes out fine, and rings my SIP office
phone just fine.  In other words, my asterisk box passes the call to my
company's server, and it goes through.  The guys at work tell me that
outside PSTN calls (like to my home PSTN phone) should work exactly the same
way (no special dialing needed, just the standard 10 digit telephone
number).  But for some reason when I dial
106 from my softphone it doesn't work.  I get a recording that tells me "the
person you are trying to reach is unavailable."

As near as I can tell (as someone unexperienced with this) from looking at
the text that gets spit out when I run "SIP debug," both calls go through
the same.  The debug info from when I dialed 106 is included below.

I know I am a dummy about this stuff.  But I am trying to learn how it
works.  Have mercy on me you experts.  Can any of you see what might be
wrong?

My guesses include two possibilities:

1.  For some reason my PSTN onramp (my own company) isn't really passing the
call to the PSTN as it should.

2.  Even for this really basic hardcoding experiment, my extensions.conf
file is too short.  For example, the connection takes longer than asterisk
expects, and so I need to tell it to keep waiting before playing the
recording.  Or something like that.

Any ideas?

Thanks,

Dave

************

INVITE sip:916xxx6000 at mysipprovider.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK032524f5
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at mysipprovider.com>
Contact: <sip:102 at 192.168.1.200>
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 15 Sep 2005 03:41:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1541 1541 IN IP4 192.168.1.200
s=session
c=IN IP4 192.168.1.200
t=0 0
m=audio 17464 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 66.81.0.87:5060
    -- Called smf-reg.sip.o1.com/916xxx6000 asterisk1*CLI>

Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK032524f5;received=24.23.48.16
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at mysipprovider.com>
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 66.81.0.87:5060 "Noisy feedback tells:  pid=22068
req_src_ip=24.23.48.16 req_src_port=5060
in_uri=sip:916xxx6000 at mysipprovider.com
out_uri=sip:916xxx6000 at 192.168.4.97 via_cnt==1"

9 headers, 0 lines
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK032524f5
Record-Route: <sip:916xxx6000 at 192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>
Record-Route: <sip:916xxx6000 at 66.81.0.87;r2=on;ftag=as0dbb6283;lr=on>
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at mysipprovider.com>;tag=as5f7d7868
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 102 INVITE
User-Agent: Asterisk SIPv2 (http://www.asterisk.org
CVS-HEAD-03/02/05-12:13:56 )
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:916xxx6000 at 192.168.4.97>
Content-Type: application/sdp
Content-Length: 210

v=0
o=root 9338 9338 IN IP4 66.81.0.97
s=session
c=IN IP4 66.81.0.97
t=0 0
m=audio 16488 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

13 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 66.81.0.97:16488 Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723),
peer - 0x1 (g723), combined -
0x1 (g723)
list_route: hop: <sip:916xxx6000 at 66.81.0.87;r2=on;ftag=as0dbb6283;lr=on>
list_route: hop: <sip:916xxx6000 at 192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>
list_route: hop: <sip:916xxx6000 at 192.168.4.97>
set_destination: Parsing
<sip:916xxx6000 at 66.81.0.87;r2=on;ftag=as0dbb6283;lr=on> for address/port to
send to
set_destination: set destination to 66.81.0.87, port 5060
Transmitting:
ACK sip:916xxx6000 at smf-reg.sip.o1.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK137a266c
Route:
<sip:916xxx6000 at 192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>,<sip:9167256000 at 19
2.168.4.97>
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at mysipprovider.com>;tag=as5f7d7868
Contact: <sip:102 at 192.168.1.200>
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 66.81.0.87:5060
    -- SIP/mysipprovider.com-9208 answered SIP/102-eb1a We're at
192.168.1.200 port 16880 Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw) Answering with non-codec
capability 0x1 (telephone-event) Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.110:5060;branch=z9hG4bK4731E35A23434B6A9D2A94DFB9D2448E
From: 102 <sip:102 at 192.168.1.200>;tag=787222251
To: <sip:106 at 192.168.1.200>;tag=as4f23aa40
Call-ID: A8B237E9-902E-4B47-979D-C0A1AECAC121 at 192.168.1.110
CSeq: 36288 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:106 at 192.168.1.200>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 1541 1541 IN IP4 192.168.1.200
s=session
c=IN IP4 192.168.1.200
t=0 0
m=audio 16880 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 192.168.1.110:5060
    -- Attempting native bridge of SIP/102-eb1a and
SIP/smf-reg.sip.o1.com-9208 asterisk1*CLI>

Sip read:
ACK sip:106 at 192.168.1.200 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.110:5060;rport;branch=z9hG4bKCC576F589D574F5FB4D8F9553C08E91A
From: 102 <sip:102 at 192.168.1.200>;tag=787222251
To: <sip:106 at 192.168.1.200>;tag=as4f23aa40
Contact: <sip:102 at 192.168.1.110:5060>
Call-ID: A8B237E9-902E-4B47-979D-C0A1AECAC121 at 192.168.1.110
CSeq: 36288 ACK
Max-Forwards: 70
Content-Length: 0

9 headers, 0 lines
asterisk1*CLI>

Sip read:
BYE sip:106 at 192.168.1.200 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.110:5060;rport;branch=z9hG4bK702E81C5C4E74BD593ABA699EB591494
From: 102 <sip:102 at 192.168.1.200>;tag=787222251
To: <sip:106 at 192.168.1.200>;tag=as4f23aa40
Contact: <sip:102 at 192.168.1.110:5060>
Call-ID: A8B237E9-902E-4B47-979D-C0A1AECAC121 at 192.168.1.110
CSeq: 36289 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0

10 headers, 0 lines
Sending to 192.168.1.110 : 5060 (non-NAT) Transmitting (no NAT):
IP/2.0 200 OK>
Via: SIP/2.0/UDP
192.168.1.110:5060;branch=z9hG4bK702E81C5C4E74BD593ABA699EB591494
From: 102 <sip:102 at 192.168.1.200>;tag=787222251
To: <sip:106 at 192.168.1.200>;tag=as4f23aa40
Call-ID: A8B237E9-902E-4B47-979D-C0A1AECAC121 at 192.168.1.110
CSeq: 36289 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:106 at 192.168.1.200>
Content-Length: 0

 to 192.168.1.110:5060
set_destination: Parsing
<sip:916xxx6000 at 66.81.0.87;r2=on;ftag=as0dbb6283;lr=on> for address/port to
send to
set_destination: set destination to 66.81.0.87, port 5060 Reliably
Transmitting:
BYE sip:916xxx6000 at 192.168.4.97 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK13e08827
Route:
<sip:916xxx6000 at 192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>,<sip:916xxx6000 at 19
2.168.4.97>
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at smf-reg.sip.o1.com>;tag=as5f7d7868
Contact: <sip:102 at 192.168.1.200>
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 66.81.0.87:5060
  == Spawn extension (from-internal, 106, 1) exited non-zero on
'SIP/102-eb1a'
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK13e08827
Record-Route: <sip:916xxx6000 at 192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>
Record-Route: <sip:916xxx6000 at 66.81.0.87;r2=on;ftag=as0dbb6283;lr=on>
From: "102" <sip:102 at 192.168.1.200>;tag=as0dbb6283
To: <sip:916xxx6000 at mysipprovider.com>;tag=as5f7d7868
Call-ID: 7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200
CSeq: 103 BYE
User-Agent: Asterisk SIPv2 (http://www.asterisk.org
CVS-HEAD-03/02/05-12:13:56 )
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:916xxx6000 at 192.168.4.97>
Content-Length: 0

12 headers, 0 lines
Destroying call '7ae2b4642cfbb601604c2d4734352e64 at 192.168.1.200'
Destroying call 'A8B237E9-902E-4B47-979D-C0A1AECAC121 at 192.168.1.110'
asterisk1*CLI>
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