[asterisk-users] Stange pause between extensions commands.

[Atis Lezdins]

On 12/14/07, Catalin S. <jonsonplayer at gmail.com> wrote:
> Hello,
>  i have a simple but annoying problem. I have the following entry in
> /etc/asterisk/externsions.conf file:
>
>  ---<Cut Here>---
>  exten => 10100,1,Wait(4)
>  exten => 10100,2,Playback(transfer,noanswer)
>  exten => 10100,3,Dial(${PHONE30},30,t)
>  exten => 10100,4,Background(extension)
>  exten => 10100,5,Background(is-curntly-unavail)

Why do you have Background() here? I think it should be Playback()

Regards,
Atis

>  exten => 10100,6,Voicemail(9999)
>  exten => 10100,7,PlayBack(vm-goodbye)
>  exten => 10100,8,Hangup
>  ---<And Here>---
>
>  Normally when i call that extension if the user is online will ring if not,
> will play: "Extension is currently unavailable" and immediately should go to
> voicemail and after voicemail will play: "Good bye" and hangup. But after
> plain "Extension is currently unavailable" is a long period of silence and
> finally will go to voicemail. On my asterisk i have the following output
> during this call:
>
>  ---<Cut Here>---
>   -- Executing [10100 at default:1] Dial("SIP/10100-082244c0", "SIP/1010|20")
> in new stack
>  [Dec 14 14:37:05] WARNING[20355]: app_dial.c:1131 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 3 - No route to destination)
>    == Everyone is busy/congested at this time (1:0/0/1)
>      -- Executing [10100 at default:2]
> BackGround("SIP/10100-082244c0", "extension") in new stack
>      -- <SIP/10100-082244c0> Playing 'extension' (language 'en')
>      -- Executing [10100 at default:3]
> BackGround("SIP/10100-082244c0", "is-curntly-unavail") in
> new stack
>      -- <SIP/10100-082244c0> Playing 'is-curntly-unavail' (language 'en')
>      -- Executing [10100 at default:4] VoiceMail("SIP/10100-082244c0", "10100")
> in new stack
>  [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
> has peer field or pending or callno (flags = 16, peer = 0x8189c00 callno =
> 0)
>  [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
> has peer field or pending or callno (flags = 16, peer = 0x82084d0 callno =
> 0)
>  [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
> has peer field or pending or callno (flags = 16, peer = 0x81e6d98 callno =
> 0)
>  [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
> has peer field or pending or callno (flags = 16, peer = 0x81daf00 callno =
> 0)
>  [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
> has peer field or pending or callno (flags = 16, peer = 0x81e4cc8 callno =
> 0)
>  [Dec 14 14:37:12] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten o
>  [Dec 14 14:37:17] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten o
>  [Dec 14 14:37:22] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten o
>  [Dec 14 14:37:27] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten o
>  [Dec 14 14:37:32] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten a
>  [Dec 14 14:37:37] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten a
>  [Dec 14 14:37:42] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten a
>  [Dec 14 14:37:47] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
> waiting for xxx:xxx at sip.xxx.com exten a
>      -- <SIP/10100-082244c0> Playing 'vm-intro' (language 'en')
>    == Spawn extension (default, 10100, 4) exited non-zero on
> 'SIP/10100-082244c0'
>  ---<And Here>---
>
>  Can anyone help me with this? I want immediately voicemail answer... maybe
> these error is the cause... I saw that in this pause the asterisk tried to
> contact this extension through my external peers (genetically named
> sip.xxx.com)... Thank you...
>
>
>
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-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835