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

2010-12-09 Thread Steve Edwards
On Thu, 9 Dec 2010, Gary Kuznitz  wrote:

> I'm getting closer.  Express Talk is now making the call.
> I'm getting an error on the cmd line:
>-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro("SIP/120-
> b6003810", "trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|") in 
> new
> stack
>-- Executing [...@macro-trunkdial-failover-0.3:1] 
> GotoIf("SIP/120-b6003810", "0?1-
> fmsetcid|1") in new stack
>-- Executing [...@macro-trunkdial-failover-0.3:2] 
> GotoIf("SIP/120-b6003810", "0?1-
> setgbobname|1") in new stack
>-- Executing [...@macro-trunkdial-failover-0.3:3] Set("SIP/120-b6003810",
> "CALLERID(num)=") in new stack
>-- Executing [...@macro-trunkdial-failover-0.3:4] 
> GotoIf("SIP/120-b6003810", "0?1-
> dial|1") in new stack
>-- Executing [...@macro-trunkdial-failover-0.3:5] Set("SIP/120-b6003810",
> "CALLERID(all)=") in new stack
>-- Executing [...@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-d...@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-gar...@192.168.168.7 for seqno 287
> (Critical Response) -- See doc/sip-retransmit.txt.

> 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(11...@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)

Does it seem odd that your console output does not match your dialplan?

I would suggest discarding PIAF or Elastix or whatever created your 
dialplan and start from scratch.

Once you master the concepts and interaction between sip.conf and 
extensions.conf you will be in a better place to evaluate the merits of 
using a GUI to create your dialplan or continue growing your own.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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

2010-12-09 Thread Gary Kuznitz
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz ) commented 
about 
[asterisk-users] (Fwd) Re:  Configuring Softphone:

> Thank you for the reply.
> 
> On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented 
> about RE: [asterisk-users] Configuring Softphone:
> 
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@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 [916618579...@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-gar...@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-gar...@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 [91myareacodepho...@dlpn_dialplan1:1] Macro("SIP/120-
b6003810", "trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|") in 
new 
stack
-- Executing [...@macro-trunkdial-failover-0.3:1] 
GotoIf("SIP/120-b6003810", "0?1-
fmsetcid|1") in new stack
-- Executing [...@macro-trunkdial-failover-0.3:2] 
GotoIf("SIP/120-b6003810", "0?1-
setgbobname|1") in new stack
-- Executing [...@macro-trunkdial-failover-0.3:3] Set("SIP/120-b6003810", 
"CALLERID(num)=") in new stack
-- Executing [...@macro-trunkdial-failover-0.3:4] 
GotoIf("SIP/120-b6003810", "0?1-
dial|1") in new stack
-- Executing [...@macro-trunkdial-failover-0.3:5] Set("SIP/120-b6003810", 
"CALLERID(all)

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

2010-12-09 Thread Gary Kuznitz
Thank you for the reply.

On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented 
about RE: [asterisk-users] Configuring Softphone:

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@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 [916618579...@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-gar...@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-gar...@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

--- End of forwarded message ---


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