Summary: | ASTERISK-16106: [regression] PlayTones() does not produce any tones in 1.6.2 (works in 1.6.1) | ||
Reporter: | Tolmino Muccitelli (tolmino) | Labels: | |
Date Opened: | 2010-05-17 06:19:32 | Date Closed: | 2011-06-07 14:05:19 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) myDebugLog | |
Description: | in CLI i see PlayTones("SIP/501-00000001", "425/200,0/200") in new stack but i don't hear nothing ****** ADDITIONAL INFORMATION ****** with asterisk 1.6.1.x that work perfectly | ||
Comments: | By: Paul Belanger (pabelanger) 2010-05-17 11:02:39 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. the specific steps or actions you took that caused you to encounter the problem, 2. the behavior you expected, and 3. the behavior you actually encountered (in as much detail as possible). This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks! By: Tolmino Muccitelli (tolmino) 2010-05-17 11:57:19 in extension.conf: [macro-callsipper] exten => s,1,set(TRANSFER_CONTEXT=standard-internal) exten => s,2,SetMusicOnHold(${ARG4}) exten => s,3,Dial(${ARG1},${ARG2},rRtTwWM(setmusiconhold,${ARG4})) exten => s,4,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,PlayBack(Music/msg_operatori_occupati) exten => s-NOANSWER,2,Wait(1) exten => s-NOANSWER,3,Hangup exten => s-BUSY,1,Playtones(425/500,0/500) exten => s-BUSY,2,Wait(3) exten => s-BUSY,3,PlayBack(call_int_BUSY) exten => s-BUSY,4,Wait(1) exten => s-BUSY,5,Hangup() exten => s-CHANUNAVAIL,1,Playtones(425/200,0/200) exten => s-CHANUNAVAIL,2,Wait(5) exten => s-CHANUNAVAIL,3,PlayBack(call_int_CHANUNAVAIL) exten => s-CHANUNAVAIL,4,Wait(1) exten => s-CHANUNAVAIL,5,Hangup() exten => s-ANSWER,1,Hangup() exten => _s-.,1,Goto(s-NOANSWER,1) [sip-phones] exten => 501,1,Macro(callsipper,SIP/501,120,501@default,papozzo) in sip.conf [Grandstream](!) ; a template type=friend host=dynamic dtmfmode=rfc2833 limitonpeers=yes disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=gsm allow=g723 allowsubscribe=yes qualify=yes call-limit=4 incominglimit=2 busy-limit=1 ;defaultip=192.168.0.60 deny=0.0.0.0/0.0.0.0 permit=192.168.1.0/255.255.255.0 [501](Grandstream) context=rc-internal callerid="Postaz. Tolm" <501> username=501 secret=********** mailbox=501@default subscribecontext=vedi-leds callgroup=3 pickupgroup=1,2,3 ..... if i call 502 that is UNAVAIABLE i hear 5 seconds of silence , and after my message "call_int_CHANUNAVAIL", but during the 5 seconds i must listen playtones (425Hz for 200ms ,0Hz for 200ms and so-on) that don't work. (i have updated that morning, and with 1.6.1.X Playtones Plays!!!) By: Paul Belanger (pabelanger) 2010-05-17 12:14:50 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: http://svn.digium.com/svn/asterisk/trun/doc/HOWTO_collect_debug_information.txt By: Paul Belanger (pabelanger) 2010-05-18 13:09:47 I suspect this is a dialplan issue, from UPGRADE.txt * SIP no longer sends the 183 progress message for early media by default. Applications requiring early media should use the progress() dialplan app to generate the progress message. By: Tolmino Muccitelli (tolmino) 2010-05-19 05:41:34 I don't think so: with my [macro-callsipper] i have a status : s-CHANUNAVAIL,1,Playtones(425/200,0/200) playtones don't send progress messages, but must send a syntetized wave at 425 Hz, like i have requested. i'm in status CHANUNAVAILABLE because instructions after Playtones are correctly done: there is a Wait of 5 seconds, and there is a message that i listen..... Problem is in Playtones synth!!! By: Paul Belanger (pabelanger) 2010-05-19 10:38:38 @Tolmino: First, we need a debug log showing the problem. http://svn.digium.com/svn/asterisk/trun/doc/HOWTO_collect_debug_information.txt 'playtones don't send progress messages', that is the problem. If you are doing early media on SIP, you now need to send 183 progress message using progress() application. Either way, collect the debug log as requested so we can see what is happening in Asterisk. Then we can triage the issue further if needed. By: Tolmino Muccitelli (tolmino) 2010-05-19 12:39:00 Hi pabelanger, i have attacched a debub log file as you request.... in that log you'll find that instruction: exten => s-CHANUNAVAIL,1,Playtones(congestion) at the place of exten => s-CHANUNAVAIL,1,Playtones(425/200,0/200) but the result is the same : i don't hear nothing!!! By: Paul Belanger (pabelanger) 2010-05-19 13:40:20 You need Progress(), since you have not answered the channel. --- exten => s-CHANUNAVAIL,1,Progress() exten => s-CHANUNAVAIL,n,Playtones(congestion) By: Tolmino Muccitelli (tolmino) 2010-05-19 13:50:14 OK.... IT WORKS!!! By: Paul Belanger (pabelanger) 2010-05-19 13:54:45 Please read UPGRADE.txt and CHANGES to confirm your Asterisk settings don't require changes before switching branches. |