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

[rnc Info Lists]

John wrote:
> 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>

<CUT out the rest of the original email..rnc>

John,
What is the dialparties.agi? You didn't mention that in your description
(or I missed it)  I have used * with 2 GS phones with no problem.

My suggestion is to go back to the simple extensions.conf file  and try it
again.  Take out all of the fancy stuff until you get the basic phone
working.

If it still doesn't work then post all relevant parts of your
extensions.conf and any changes you made in sip.conf along with the trace.

My GS SIP.conf for one of the phones is:
[2001]
type=friend
username=2001
secret=test2
host=dynamic
context=local-extensions   <--this will probably be different in your setup

Extension.conf for ringing that phone is:
exten => 2001,1,Dial(SIP/2001,20,Ttr)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

They probably aren't perfect but they do work.

Robert