Catalin S.
2007-Dec-14 12:49 UTC
[asterisk-users] Stange pause between extensions commands.
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) 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... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071214/0f6f5470/attachment-0001.htm
Atis Lezdins
2007-Dec-14 13:13 UTC
[asterisk-users] Stange pause between extensions commands.
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... > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835
Tony Plack
2007-Dec-14 17:08 UTC
[asterisk-users] Stange pause between extensions commands.
> /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) > exten => 10100,6,Voicemail(9999) > exten => 10100,7,PlayBack(vm-goodbye) > exten => 10100,8,Hangup > ---<And Here>--- > > > ---<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>--- >I don't mean to be picky but something isn't right here. 1. The priority numbers in the log file do not match to the snippet from extensions.conf. I assume you made changes to the extension.conf after this file was created. 2. It seems that you are trying to dial exten 10100 from SIP/10100 and it is redirected to SIP/1010. No issues there, just noting it. 3. The call is transferred to the voice mail module and the delay is occurring because of the configuration of this module The call is trying to reach an operator ( exten o) The call is trying to reach exten a Then the call is moving to standard voicemail Please review your config in extensions.conf and find an area which lists extension o. This is the operator extension and I assume that you have the operator=yes in voicemail.conf. I would look to these sections to determine what is occurring. Somewhere in your config you have the system trying to call xxx:xxx at sip.xxx.com for the operator. If you still need help, please post the general and specific settings from voicemail.conf for review.