similar to: SIP and music on hold

Hello: I have read that music on hold requires a timing source (which I never had to worry about previously since the server had zaptel hardware in it)...now I'm configuring a server in a colo which has no zaptel hardware. If I use the dialplan to run MusicOnHold(), I do get the music upon dialling that extension, but if I try to use the hold button on either a 7960 or X-Lite I get

I have a modem on a port of a channel bank dialing out a FXO port to the PSTN. It wont train at any speed. If I put another modem on a another port of the channel bank, it will connect to it, but at @31Kb. Anybody try this and have it work ?

Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the "other end" a diff type of trunk ie: 7960sip --> asterisk --> IAX2 --> PRI 7960sip -->

Ok, this falls under the newbie category: Has anybody created any documentation on using musiconhold or call parking? I have found sample config files for musiconhold, but I'm not sure how they work. [musiconhold.conf] [classes] loud=>mp3:/var/lib/asteriks/mohmp3 How do I then set up this feature in extensions.conf? I can't seem to find an example of what I'm looking for (or I

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

Thanks to everyone that responded to my channel bank question. Ive decided that the Adit 600 would be a good choice. Then I got to thinking about SIP phones and wondered if their quality has progressed to the point that they can be deployed to customers who "just want their phones to work" and wouldn't tolerate any SIP hickups. As for pricing, I would think the SIP phones would

I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Hi All, I have what IMHO is an interesting issue. I'm using Cisco 7940's with the 7.2 SIP load and Asterisk CVS-HEAD-09/10/04-10:11:46. Everything is working great so far, except one small issue. When a user presses the 'Trnsfer' soft-key, dials the other extension, and presses 'Trnsfer' again, before the other party picks up, hold music for the original caller

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

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 --

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/59c113df/attachment.htm -------------- next part -------------- Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card,

Does anyone know how to maximize music on hold quality on calls inbound from PSTN? I know that it is common to have choppy and static sounding music on hold when connecting via PSTN but how can that be minimized? I assume that the bitrates, type of music, etc can minimize the effects. Does anyone have any experience in this area? Do you know where I should look for more information?

This is probably going to sound really silly and I must be confused about it. Maybe someone can set me straight. I've been tinkering for a while with * and a number of different FXO/FXS cards, SIP phones, and ATAs trying to get a feel for what works and what doesn't. In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and now an SPA-841. Each of these allow me to configure at

I have noticed a problem with the Cisco 7940/7960 phones where if you put your voice-mail box on hold using soft keys and come back you can no longer navigate. I am curious if anyone else can duplicate this problem. Happens reliably for me with the 7940 phones. I use rfc2833 for DTMF. I would think it was a Cisco bug, but for the fact that this did not happen with older version of

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

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 have a requirement to play different hold messages depending upon the extension that originated the call. I noticed a musicclass setting in sip.conf, but it appears this is global. I tried setting this on all of my individual extensions, but it didn't have any affect. Is there a way to achieve this, either through sip.conf or in the dial plan? Thanks, James

Does IAX support music on hold? It seems only my SIP phones do. Is this correct?

Is there a clean way to disable music on hold for a specific user sip user? I have seen one example that creates a class called [none] that points to an empty directory, which creates log errors that are annoying (but not harmful?) -------------- next part -------------- An HTML attachment was scrubbed... URL: