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2004 May 29
4
PlayTones problem
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the con...
2009 Sep 07
2
Echo and Playtones not working on SIP after upgrade
Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten => 1003,1,Wait(1) exten => 1003,n,Playtones(!1050/1000) exten => 1003,n,Wait(1) exten => 1003,n,StopPlaytones exten => 1003,n,Echo exten => 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP c...
2003 Jun 12
0
Playtones unexpected hangups
1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I can provide more...
2005 Jul 27
0
Playtones not passing sound to incoming SIP connection
...lling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any sound to the incoming SIP connection. I have added the following to my [incoming_sipgate] context (which has two other sipgate numbers in there which both work for incoming and outgoing calls), and on the console I can see all the lines being executed. exten...
2014 Oct 30
1
PlayTones not working
I?m trying to use Playtones to have a tone played periodically throughout phone calls. Unfortunately, I can?t seem to get PlayTones to work. I never hear the audio tones. Here is the output on the Asterisk console. -- Executing [19525553312 at proxy-dial:2] PlayTones("SIP/testphone-00000032", "1400/500,200...
2014 Oct 31
1
PlayTones while in call
I?ve gotten PlayTones to work, however it stops playing the tones as soon as the call is answered. I would like to use PlayTones during the call because I want to have a tone/beep played in the background while call recording is going on. Anyone know a way to get PlayTones to work while call is in progress? Alterna...
2019 Jan 31
2
Dailplan with playtones
Hello I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangup() All is fine. Hello world is Playback and I hear a ring tone. If I remove the Playback hello-world. No ring tone is hearing anymore. Also not when my softphone is ring. What I want is: - Play caller a ring tone when softphone...
2004 Nov 25
1
Can't hear playtones?
...nel = sip && dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. I've changed the county setting to NL in indications.conf and created this test extension: Exten => s,1, answer Exten => s, 2, playback(test) Exten => s, 3, playtones(busy) But I can't hear a busy tone on my sip phone, the call is answered, I hear the test file playback, but no busy tone. I tried to enter the values directly into playtones, but that didn't work either. Am I missing something? Kind regards, E. Versaevel -------------- nex...
2009 May 27
1
Playtones Volume
I've researched my brains out on this, and can't find any answer. Is there a way to adjust the level of the tones generated through the Playtones command? I'm thinking that I may have been approaching this incorrectly by targeting indications.conf since the tones are being called via the Playtones application. My sense is that it's not possible due to the lack of response from the way I was approaching the problem initially....
2019 Jan 31
2
Dailplan with playtones
...basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten => o2,n,Dial(SIP/10&SIP/20&Local/s at no-op,25,rt) >> exten => o2,n,Playtones(425/1000,0/4000) >> exten => o2,n,Wait(30) >> exten => o2,n,Hangup() >> >> All is fine. Hello world is Playback and I hear a ring tone. > > That seems a most odd thing to want the caller to hear. > >> If I remove the Playback hello-world. No ring ton...
2005 Jul 26
1
Are busy and congestion behaving differently than documented?
...n. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer exten => 7777,2,Playtones(busy) exten => 7777,2,busy(35) exten => 7777,3,Hangup In both cases I seem to be getting the EXACT behaviour. Asterisk plays a 'busy' signal for 35 seconds I have also tried this with congestion (instead of busy). What is strange is that in either case, the busy tones are coming...
2015 May 09
2
No application 'Playtones'
Hello Everyone, We have most of the modules commented out. Can someone please let me know which modules needed to be included for Playtones? Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150509/15ea3418/attachment.html>
2005 Aug 26
1
bridging sip to capi, no playtones back to caller
I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880", "CAPI/02myisdnnum:347callednum") in new stack -- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0x193 -- Called 02myisdnnum:347callednum...
2015 May 11
0
No application 'Playtones'
symack wrote: > Hello Everyone, > > We have most of the modules commented out. Can someone please let me > know which modules needed to be included for Playtones? The PlayTones application is in the app_playtones module. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
2004 Oct 05
0
sipura 3000 , music on hold (playtones)
hi, I have some problem with musiconhold or playtones (background,...) in this context someone dial out thru sipura 3000: Executing Dial("Zap/1-1", "SIP/sipura3000/054419949|20|m") in new stack -- Called sipura3000/054419949 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/sipura3000-61fe is ring...
2004 May 18
2
registering in sipphone
...but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^ exten => _X.,2,Playtones(congestion) exten => _X.,102,Playtones(busy) exten => h,1,Hangup randy
2004 Jul 25
1
how do I play congestion tone when Zap channels are full?
...w do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten => _9NXXXXXX,2,Congestion exten => _9NXXXXXX,3,Playtones(congestion) exten => _9NXXXXXX,102,Busy exten => _9NXXXXXX,103,Playtones(busy)     __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail
2004 Sep 27
0
chan_sip.c 183 / 180 handling, unexpected results & playtone bug ...
Guys, I am currently suffering the exact same bug as described in BUG ID 0001959, in summary, receiving a 'Asked to transmit frame type 64, while native formats is 8 (read/write = 8/8)' when call connects then hangs up. It appears to be caused by the playtones_generator routine in indications.c trying to produce a ring signal on the channel. The reason I am hitting this problem seems to be that the SIP UA I am using is sending a 183 Session Progress prior to a 180 Alerting. It appears to me that Asterisk is treating the 183 similar to a Q.931 Call Pro...
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in
2006 Jan 22
0
Interrupting ring to go to voicemail pickup -- How to ring after Answer()?
...irst answer the phone. The trouble is then providing a ring tone to the calling party. I stumbled upon the solution below, which seems to work. But there are a number of things I don't understand, which makes the whole thing seem really brittle. 1) Why does this not work if you leave off the Playtones() call? I thought Dial() was supposed to automagically indicate ringing and provide the requsite sounds? 2) Why is the r flag reqired in the Dial() call? If I take it out I don't hear any ringing even though Playtones should be playing. 3) If I take the r flag out of Dial() and put a call...