[asterisk-users] (Fwd) Re: Configuring Softphone
Steve Edwards
asterisk.org at sedwards.com
Fri Dec 10 04:56:25 UTC 2010
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 [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 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)
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 sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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