[Asterisk-Users] Newbie - getting two local phones to communicate would be a good start :)

Fri Jan 2 15:57:28 MST 2004

Hi

This is hard work :) I have read the Asterisk Handbook, BudgeTone User
Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages
and more.

I am not a linux newbie but am new to Asterisk. I have failed to find any
docs that explain how to get a very very simple, minimal, system up and I am
trying to get the following to work:

2 BudgePhone 102D connected on a LAN to a linux RH9 server running Asterisk
IP 10.0.1.198 - I just want to be able to dial from one phone and talk to
the other :) I have another phone connected to FWD sucesfully and the LAN is
NATed at the PC that is acting as the Asteriski server and firewall. But for
now its just two phones on a LAN - I'll conquer FWD and IAX later....

The extensions are 5702 and 5703. I can "dial" direct from one phone to the
other (not using Asterisk) and the other one rings and answers fine with a
voice path.

When I dial "5703" from 5702 (thus via asterisk), 5703 rings. If I take it
off hook it stops ringing but I can still hear ringing on 5702. After a few
seconds I get the "rapid-beep" tone on both phones. No voice.

I get this from asterisk CLI

*CLI>
*CLI>
    -- Executing Macro("SIP/5702-a5be", "exten-aa|5703") in new stack
    -- Executing SetMusicOnHold("SIP/5702-a5be", "random") in new stack
    -- Executing Macro("SIP/5702-a5be", "dial|20|tr|5703") in new stack
    -- Executing AGI("SIP/5702-a5be", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Caller ID name is 'John workroom #1' number is '5702'
    --  dialparties.agi: Added extension 5703 to extension map
    --  dialparties.agi: Extension 5703 cf is disabled
    --  dialparties.agi: Extension 5703 do not disturb is disabled
    --  dialparties.agi: DbSet CALLTRACE/5703 to 5702
  dialparties.agi: About to execute Dial(SIP/5703|20|tr)
    -- AGI Script Executing Application: (Dial) Options: (SIP/5703|20|tr)
    -- Called 5703
    -- SIP/5703-5fdc is ringing
    -- SIP/5703-5fdc answered SIP/5702-a5be
    -- Attempting native bridge of SIP/5702-a5be and SIP/5703-5fdc
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call c9ab05e4-254e-b34f-7d6c-067b675fa15d at 10.0.1.202 for seqno
36119 (Response)
  == Spawn extension (macro-dial, s, 1) exited non-zero on 'SIP/5702-a5be'
in macro 'dial'
  == Spawn extension (macro-exten-aa, s, 2) exited non-zero on
'SIP/5702-a5be' in macro 'exten-aa'
  == Spawn extension (johnhome, s, 1) exited non-zero on 'SIP/5702-a5be'

*CLI>
*CLI>

I've turned on SIP debug but can not see any errors reported. This look like
the moment of failure:

Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202
From: "John Coll 5702"
<sip:5702 at 10.0.1.198;user=phone>;tag=bfbd6f17-1d79-ed6b-1710-239de5724559
To: <sip:5703 at 10.0.1.198;user=phone>;tag=as3835ce1f
Call-ID: d3cb51f8-4d0a-8d70-bb8a-68986f1754bb at 10.0.1.202
CSeq: 28108 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5703 at 10.0.1.198>
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 27210 27211 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18922 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

 to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call d3cb51f8-4d0a-8d70-bb8a-68986f1754bb at 10.0.1.202 for seqno
28108 (Response)

The Grandstream phones are configured like this:

Login password	xxx
MAC	00.0B.82.00.4B.57
IP	10.0.1.202
Subnet	255.255.255.0
Default router	10.0.1.198
DNS server #1	10.0.1.198
DNS Server #2	158.152.1.43

SIP Server:	10.0.1.198
Outbound Proxy:
SIP User ID: 	5702
Authenticate ID: 	5702
Authenticate Password: 	xxx  (same if this is set to an empty string)
Name: 	John Coll 5703
Timezone	GMT
SIP User ID is
phone number:	yes

And sip.conf contains this

[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
externip = 10.0.1.198   ; Addres

[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid="John workroom #1" <5702>
mailbox=5702
nat=no

[5703] is similar

extensions.conf is currently slightly modified verson of Zac Sprackett's
file http://sprackett.com/asterisk/ - its a bit long so I won't paste yet.
However I have had the same result with a much simpler extensions.conf -
some days ago.

 Any help would really be appreciated as I am stuck and finding the process
hard because I can't seem to find a basic introduction aimed at getting me
up and running with the most basic of systems. Perhaps you can point me to a
BASIC and minimal set of configuration files for example for a SIP phone or
two on a NAT LAN with an X100P plugged into PSTN. I guess that is where most
people start - or should I start somewhere else?

I've been at this off and on for two weeks ....   Linux admin and firewalls
seem trivial compared to this so I must be missing something pretty basic :)

thanks

john