similar to: music on hold skipping

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 -> 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts

I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom "Hold" button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays.

Hi, Happy Thanksgiving to those of us in the USA... Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to this snippet of dialplan: exten => 999900,1,DeadAGI(pq.agi,999950) pq.agi then plays a prompt (which I hear just fine): [Nov 26 02:42:47] VERBOSE[28721] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/pq.agi [Nov 26 02:42:47] VERBOSE[28721]

Hello Yesterday, i brought linksys PAP2 and have success with that. The only thing that does not go well is the music on hold. When i press 'hold' button from the telephone set instead of playing the music on hold that i have setup in Asterisk, Telephone Set plays its own MOH. Is there any way to tackle this issue. Regards Taimur Hasan -THQ- !!!ONE

Hey all; I am experiencing an issue with music on Hold. I am running asterisk version 1.4.22, and have a default script set up in two places for MoH playback. For internal devices to my network that are SIP peering with asterisk, they simply dial 123 and hear the MoH music immediately. I'm using the default setup, where it just plays the wave files in the /var/lib/asterisk/moh directory. I

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Hello I have a rather limit hardware where i try to run Asterisk, 1 GHz VIA C3. Everything works fine except music playback on MOH. I have encoded the music in same codac as as the voicestream (ulaw), ulaw is used end to end. MOH fades in and out with varius volume, and is very choppy. What I don't quite understand is when I put the music file as "you are first in line" it

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

A feature we're often asked for in our ITSP product is to allow customers to upload their own music on hold, or to have it recorded for them by a recording studio with the latest news, weather, etc, punctuated by "Welcome to <customer>, please hold". Since there may be thousands or tens of thousands of customers, and perhaps 10% of customers may want this feature with a

I've got the same problem. MusicOnHold works if I use something like: Exten => 1111,1,MusicOnHold() but if I try to answer a call and then transfer or put on hold the call, I get no music. Does anyone have any idea? Bye, Gianluca. _____ Da: Kanishka Somaratne [mailto:kani@technoportal.biz] Inviato: gioved? 17 marzo 2005 5.53 A: asterisk-users@lists.digium.com

Has anyone encountered this SIP problem? I dial a SIP phone from a ZAP phone and after talking for a few minutes the SIP user transfers the ZAP user to another ZAP user. The transfer works correctly, but when the native bridge occurs -- Attempting native bridge of Zap/22-1 and Zap/16-1 the music on hold never stops. The two lines are bridged correctly and you can hear one another but you

I'm using the default mp3 files that ship with Asterisk: fpm-calm-river.mp3 fpm-world-mix.mp3 If they were variable bit rate, I think I would see a warning about 'varibel' or similar...is anyone else able to get these files to work? -Mike- ________________________________ From: Sander [mailto:crombeen@rommelweb.nl] Sent: Tuesday, May 17, 2005 6:34 PM To: 'Asterisk

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983]

Hi list, I'm planning and testing a distributed asterisk deployment throughout several sites; each will be connected to the PSTN and all of them among themselves via IAX trunks. Phones will be SIP. I guess I already "solved" (worked-around, actually) asterisk's codec negotiation limitations regarding local G.711 utilization vs. remote G.729 while minimizing

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

Thanks Ben, Adam and Petr for the feedback! So currently things that need to be done for the Monitor resource are: 1. Name files uniquely. Adam, your naming suggestion is great. I think we should stick with that, with a minor change: I don't think we should put destination channel name in the file names. In some instances there will be no destination channels (plain IVR: play, record, dtmf),