[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



More information about the asterisk-users mailing list