Hello and thank you for reply... I tried with Playback() and is the
same effect. Is curious because sometime there's no pause other time is
a long pause.<br>
<br>
Anybody have other idea?<br>
<br>
Thank you.<br><br><div><span class="gmail_quote">On 12/14/07, <b class="gmail_sendername">Atis Lezdins</b> <<a href="mailto:atis@iq-labs.net">atis@iq-labs.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On 12/14/07, Catalin S. <<a href="mailto:jonsonplayer@gmail.com">jonsonplayer@gmail.com</a>> wrote:<br>> Hello,<br>> i have a simple but annoying problem. I have the following entry in<br>> /etc/asterisk/externsions.conf file:
<br>><br>> ---<Cut Here>---<br>> exten => 10100,1,Wait(4)<br>> exten => 10100,2,Playback(transfer,noanswer)<br>> exten => 10100,3,Dial(${PHONE30},30,t)<br>> exten => 10100,4,Background(extension)
<br>> exten => 10100,5,Background(is-curntly-unavail)<br><br>Why do you have Background() here? I think it should be Playback()<br><br>Regards,<br>Atis<br><br>> exten => 10100,6,Voicemail(9999)<br>> exten => 10100,7,PlayBack(vm-goodbye)
<br>> exten => 10100,8,Hangup<br>> ---<And Here>---<br>><br>> Normally when i call that extension if the user is online will ring if not,<br>> will play: "Extension is currently unavailable" and immediately should go to
<br>> voicemail and after voicemail will play: "Good bye" and hangup. But after<br>> plain "Extension is currently unavailable" is a long period of silence and<br>> finally will go to voicemail. On my asterisk i have the following output
<br>> during this call:<br>><br>> ---<Cut Here>---<br>> -- Executing [10100@default:1] Dial("SIP/10100-082244c0", "SIP/1010|20")<br>> in new stack<br>> [Dec 14 14:37:05] WARNING[20355]: app_dial.c:1131 dial_exec_full: Unable to
<br>> create channel of type 'SIP' (cause 3 - No route to destination)<br>> == Everyone is busy/congested at this time (1:0/0/1)<br>> -- Executing [10100@default:2]<br>> BackGround("SIP/10100-082244c0", "extension") in new stack
<br>> -- <SIP/10100-082244c0> Playing 'extension' (language 'en')<br>> -- Executing [10100@default:3]<br>> BackGround("SIP/10100-082244c0", "is-curntly-unavail") in
<br>> new stack<br>> -- <SIP/10100-082244c0> Playing 'is-curntly-unavail' (language 'en')<br>> -- Executing [10100@default:4] VoiceMail("SIP/10100-082244c0", "10100")
<br>> in new stack<br>> [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still<br>> has peer field or pending or callno (flags = 16, peer = 0x8189c00 callno =<br>> 0)<br>> [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
<br>> has peer field or pending or callno (flags = 16, peer = 0x82084d0 callno =<br>> 0)<br>> [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still<br>> has peer field or pending or callno (flags = 16, peer = 0x81e6d98 callno =
<br>> 0)<br>> [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still<br>> has peer field or pending or callno (flags = 16, peer = 0x81daf00 callno =<br>> 0)<br>> [Dec 14 14:37:07] WARNING[20355]: chan_iax2.c:10020 find_cache: DP still
<br>> has peer field or pending or callno (flags = 16, peer = 0x81e4cc8 callno =<br>> 0)<br>> [Dec 14 14:37:12] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">
xxx:xxx@sip.xxx.com</a> exten o<br>> [Dec 14 14:37:17] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">xxx:xxx@sip.xxx.com</a> exten o<br>> [Dec 14 14:37:22] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">xxx:xxx@sip.xxx.com</a> exten o<br>> [Dec 14 14:37:27] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">
xxx:xxx@sip.xxx.com</a> exten o<br>> [Dec 14 14:37:32] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">xxx:xxx@sip.xxx.com</a> exten a<br>> [Dec 14 14:37:37] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout
<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">xxx:xxx@sip.xxx.com</a> exten a<br>> [Dec 14 14:37:42] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">
xxx:xxx@sip.xxx.com</a> exten a<br>> [Dec 14 14:37:47] WARNING[20355]: chan_iax2.c:10104 find_cache: Timeout<br>> waiting for <a href="mailto:xxx:xxx@sip.xxx.com">xxx:xxx@sip.xxx.com</a> exten a<br>> -- <SIP/10100-082244c0> Playing 'vm-intro' (language 'en')
<br>> == Spawn extension (default, 10100, 4) exited non-zero on<br>> 'SIP/10100-082244c0'<br>> ---<And Here>---<br>><br>> Can anyone help me with this? I want immediately voicemail answer... maybe
<br>> these error is the cause... I saw that in this pause the asterisk tried to<br>> contact this extension through my external peers (genetically named<br>> <a href="http://sip.xxx.com">sip.xxx.com</a>)... Thank you...
<br>><br>><br>><br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>><br>> asterisk-users mailing list
<br>> To UNSUBSCRIBE or update options visit:<br>><br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br><br><br>--<br>Atis Lezdins
<br>VoIP Developer,<br>IQ Labs Inc.<br><a href="mailto:atis@iq-labs.net">atis@iq-labs.net</a><br>Skype: atis.lezdins<br>Cell Phone: +371 28806004<br>Work phone: +1 800 7502835<br><br>_______________________________________________
<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>