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

[David Gomillion]

Just to second this post: I had the same symptoms and resolved them by
tweaking my firewall.

Hope this helps,
David Gomillion

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of SW
Sent: Friday, January 02, 2004 11:07 PM
To: "John Coll"; asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Newbie - getting two local phones to
communicate would be a good start :)

Hi John,

If your effort is to make calls between two GS phones via *, here is
what
you need.
You need all three devices in the same LAN, so set both phones and * to
10.0.1.98/24.

After that from your asterisk Linux box ping both phones. If that is
successful you know your layer 1,2 and 3 are ok. Disable all fire-walls,
iptables, ipchains in Linux box.

Now in * you need two files in /etc/asterisk

sip.conf and extensions.conf.

rename or delete both those existing files.

Here are the minimum you probably need in these two files.

sip.conf :

[general]
port=5060
allow=all
maxexpirey=180
defaultexpirey=160

[5702]
type=friend
username=5702
context=internal
dtmfmode=info

[5703]
type=friend
username=5703
context=internal
dtmfmode=info


And extensions.conf

[internal]

exten => _57XX,1,Dial(SIP/${EXTEN})


Save both files and issue command reload from * CLI.

Now you should be able to call from one phone to another.

while making calls enable sip debug and study the messages going in and
out.
Also if you have ethereal fire that up and capture SIP packets. and see
how
the SIP negotiation goes on. This will help you when you start moving to
fwd, IAXTEL etc. etc.

good luck.

SW




From: "John Coll" <john.coll at csoft.co.uk>
To: <asterisk-users at lists.digium.com>
Date: Fri, 2 Jan 2004 22:57:28 -0000
Subject: [Asterisk-Users] Newbie - getting two local  phones to
communicate
would be a good start :)
Reply-To: asterisk-users at lists.digium.com

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-1710-239de572455
9
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)


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