[Asterisk-Users] Starting From Scratch

Wed Sep 14 20:54:53 MST 2005

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 192.168.4.97>
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 192.168.4.97>
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>