similar to: Music on hold not initiating RTP stream?

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

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

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to

Hi, I've seen some postings on the Directory application, but haven't seen too many resolution postings. Has anyone experienced where the Directory app doesn't even answer when called? For example, using the config below, dialing 899 results in just a continual ringing sound. extensions.conf exten => 899,1,Directory(local) exten => 899,2,Hangup [local] exten =>

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/9e69a5c9/attachment.htm>

I'm running Connected to Asterisk 1.4.18-1. We have a T1 PRI and a SIP trunk coming in from our IVR. When people call in on the SIP trunk, I experience the following problem: I put the caller on hold, MOH starts. MOH quits after approx ten seconds. When I pick the call back up off of hold, they can't hear me for about 20 seconds (sometimes longer or not at all) but I can hear them. This

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this

Hi there, We're experiencing the following problem on a handful of boxes and I'm wondering if anyone has any pointers for me. First up, hardware configs on boxes are the same: Dell 860, Dual Core Xeon, 1Gb RAM, 2x 160Gb SATA's in Software RAID 1. Various Trixbox versions - 2.2.3 and 2.4.2 to name two of them. Problem is that occasionally the music on hold speeds right up -

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

Anybody have this work? If I dial the extension from an analog connection, it works. From a Cisco 7960 it doesn't.

Is any way to limit music on hold (or ringtones) to specified time ? I need it to play it ~ for 7 seconds . How to do this ? in dial plan i have: exten => _021XXXXXX,4,Dial,Zap/1/BYEXTENSION||r when go to this extension it rings once! and then asterisk say : -- Zap/1-1 answered Modem[i4l]/ttyI0 and it stop ringing ;) becouse mean that other end is ringing :) .. BUT when the other

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

what can be wrong ? -- Executing MusicOnHold("CAPI[contr1/8]", "default") in new stack WARNING[13326]: File res_musiconhold.c, Line 259 (moh0_exec): Unable to start music on hold (class 'default') on channel CAPI[contr1/8] == Spawn extension (remote, s, 6) exited non-zero on 'CAPI[contr1/8]' thomas

Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full

I've only gotten to hear Music On Hold once. I am running the right version of mpg123 .059r as downloaded from mpg123.de and compiled locally So I'm looking for any help on getting this up and running. I can see on the console that the SIP phone is placing the call on hold, but there is no audio coming out on the other phone. Mucho, thanks john

I'm curious if anyone has used a radio for MOH? If so, how did you set it up? I have a client who is interested in using a radio for the music on hold, since that is what they did with their old phone system. Thanks, Leif Madsen.