[asterisk-users] (Fwd) Re: Configuring Softphone

[Gary Kuznitz]

On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz <docfxit at theoffice.la>) commented about 
[asterisk-users] (Fwd) Re:  Configuring Softphone:

> Thank you for the reply.
> 
> On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented 
> about RE: [asterisk-users] Configuring Softphone:
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz 
> > Sent: Wednesday, December 08, 2010 1:27 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Configuring Softphone
> > 
> > The phone is finally registering.   That's great.
> > 
> > I'm trying to understand what all these lines in Extensions.conf are
> > defining.
> > I can't make heads or tails them.  I have been reading the manual 
> > AsteriskManualTheFutureOfTelephony2ndEdition.
> > 
> > I'm currently getting an error when placing a call on the cmd line saying:
> > NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to 
> > extension '91AreaCodePhone#' rejected because extension not found.
> >  
> > 
> > What I have in Extensions.conf is:
> > [gary-incomming]
> > exten => 1001,1,Dial(IAX2/gogh)
> > exten => 1001,2,HangUp()
> > exten => 120,1,Dial(SIP/Gary)
> > exten => Gary,1,Goto(120,1)
> > exten => i,1,Playback(invalid)
> > exten => i,2,Goto(s,1)
> > exten => s,1,Wait(1)
> > exten => s,2,Answer()
> > exten => s,3,NoOp(${CALLERID})
> > exten => s,4,NoOp(${CALLERIDNUM})
> > exten => s,5,NoOp(${CALLERIDNAME})
> > exten => s,6,Wait(4)
> > exten => s,7,Playback(vm-goodbye)
> > exten => s,8,Wait(2)
> > exten => s,9,HangUp() 
> > 
> > What I have in Sip.conf is:
> > [authentication]
> > 
> > [general]
> > context = default
> > allowoverlap = no
> > bindport = 5060
> > bindaddr = 0.0.0.0
> > srvlookup = yes
> > limitonpeers = yes
> > allowguest=no
> > nat=yes         
> > 
> > [Gary]
> > type = friend
> > username = Gary
> > callerid = 120
> > secret = password
> > host = dynamic
> > defaultip = dynamic
> > context = gary-incomming
> > dtmfmode = rfc2833
> > allow=all  
> > 
> > Frustrated,
> > 
> > Gary
> > 
> > Without any other comment, you need 
> > exten => _91.,1,Dial(DAHDI/g1/${EXTEN})
> > in the gary-incomming context.
> > 
> > As defined now, Gary can 
> > #1 answer a call
> > #2 call IAX/gogh using 1001
> > 
> 
> I entered the exten line you suggested:
> [gary-incomming]
> exten => _91.,1,Dial(DAHDI/g1/${EXTEN})
> 
> I removed all other lines in [gary-incomming]
> 
> When I place a call I get on the cmd line:
>      -- Executing [916618579191 at gary-incomming:1] Dial("SIP/Gary-08941b20", 
> "DAHDI/g1/916618579191") in new stack
>     -- Called g1/916618579191
>     -- DAHDI/1-1 answered SIP/Gary-08941b20
> [Dec  9 14:00:37] WARNING[5630]: chan_sip.c:1958 retrans_pkt: Maximum retries 
> exceeded on transmission 1291829914-5076-GARYLT at 192.168.168.7 for seqno 669 
> (Critical Response) -- See doc/sip-retransmit.txt.
> [Dec  9 14:00:37] WARNING[5630]: chan_sip.c:1980 retrans_pkt: Hanging up call 
> 1291829914-5076-GARYLT at 192.168.168.7 - no reply to our critical packet (see 
> doc/sip-retransmit.txt).
>     -- Hungup 'DAHDI/1-1'
>   == Spawn extension (gary-incomming, 916618579191, 1) exited non-zero on 
> 'SIP/Gary-08941b20'
> 
> Do you have any ideas?  Would you like to see what is in extensions.conf for a local 
> extension?
> 
> Thank you,
> 
> Gary

I'm getting closer.  Express Talk is now making the call.
I'm getting an error on the cmd line:
    -- Executing [91MyAreaCodePhone#@DLPN_DialPlan1:1] Macro("SIP/120-
b6003810", "trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|") in new 
stack
    -- Executing [s at macro-trunkdial-failover-0.3:1] GotoIf("SIP/120-b6003810", "0?1-
fmsetcid|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:2] GotoIf("SIP/120-b6003810", "0?1-
setgbobname|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:3] Set("SIP/120-b6003810", 
"CALLERID(num)=") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:4] GotoIf("SIP/120-b6003810", "0?1-
dial|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:5] Set("SIP/120-b6003810", 
"CALLERID(all)=") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:6] Goto("SIP/120-b6003810", "1-
dial|1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial at macro-trunkdial-failover-0.3:1] Dial("SIP/120-b6003810", 
"Dahdi/g1/1MyAreaCodePhone#") in new stack
    -- Called g1/1MyAreaCodePhone#
    -- DAHDI/1-1 answered SIP/120-b6003810
    -- Hungup 'DAHDI/1-1'
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 
'SIP/120-b6003810' in macro 'trunkdial-failover-0.3'
  == Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 
'SIP/120-b6003810'
[Dec  9 18:40:39] WARNING[5806]: chan_sip.c:1958 retrans_pkt: Maximum retries 
exceeded on transmission 1291829922-5076-GARYLT at 192.168.168.7 for seqno 287 
(Critical Response) -- See doc/sip-retransmit.txt.

I don't know if this has anything to do with Express Talk using Local RTP ports to 
listen 8000-8020 and Asterisk using 10000 and up.  I tried changing Express Talk to 
10000-10020 and forwarding those to Asterisk.  It didn't seam to help.

I currently have in extensions.conf:
[gary-incomming]
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,NoOp(${CALLERID})
exten => s,n,NoOp(${CALLERIDNUM})
exten => s,n,NoOp(${CALLERIDNAME})
exten => s,n,Wait(4)
exten => s,n,Playback(tt-weasels)
exten => s,n,Voicemail(11111 at vm-test)
exten => s,n,Wait(2)
exten => s,n,Playback(vm-goodbye)
exten => s,n,Wait(2)
exten => s,n,HandUp()

exten => 120,1,Dial(SIP/gary)
exten => gary,1,Goto(120,1)

exten => i,1,Playback(invalid)
exten => i,2,Goto(s,1)

There are some other issues but I thought I should pose one question at a time.

Could someone please give me an idea as to why I'm getting the warning?

Thank you,

Gary