similar to: no audio for ring group.

Hi, Apologies if this has been asked before, but I don't seem to be able to find any info on it anywhere. Sometimes when placing a call on hold, the caller hears digitized/ robotic music on hold that gradually improves over the course of about 20 - 30 seconds until it sounds pretty normal. The first time the call is placed on hold the music sounds normal. If that same call is

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call

G'day all, I've been googling myself silly looking for help on this one but have come up blank. I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the

Shane Wegner wrote: > Hello all, > > I have two handsets connected to FXS ports on a TDM400P, > both GE models but one rings and the other does not. The > phone models are not identical. The phone which doesn't > ring on the TDM does ring when connected to a regular POTS > line and I tried connecting another phone to the port and > it rings fine. Do you have the

Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

Hi everyone, This is my first post to the list, although I am a long term user of Asterisk. I have recently found a problem that I just can't seem to solve. I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single

>The call always has echo on it. The Asterisk sip extension >hears them selves echoing. The remote party does not notice any >difference. >I have also notices that the Asterisk >levels are very hot from T1-PRI to Sip. I have had good success fiddling with the txgain and rxgain values in zapata.conf on my PRI. In my setup, cranking the gain down a LOT eliminated most of the

Hi all, I'm getting mired in conflicting and dated information, so I think I need to ask this of live humans... :-) I'm essentially trying to create an Ogg file that works like DVD video does. In particular, in addition to the Theora video stream, there will be multiple audio "tracks" (By "track", I mean a collection of channels to be played all at once, but only one

I have a system running CVS HEAD 6/30/2004. We've only been using it for PSTN to channel bank handsets, but have decided to add sip phones into the mix. Now I have quite a few systems running sip phones just fine as well as some running both sip and analog via channel banks or tdm cards. When we tried to set up some sip extensions (they are behind nats, we are using xten light, and have

I have ACL's enabled and am getting a new error, in the Samba log (V 3.0.1Pre1, when attempting to set permissions on a file through Win2000: get_domain_user_groups: primary gid of user [terry] is not a Domain group ! get_domain_user_groups: You should fix it, NT doesn't like that Do I need to create a group on the windows(2000) side? The entries in the domaingroup.map

Currently, I have a link_to_remote like this: <%= link_to_remote("Get Results", :update => ''query_results'', :with => "''tags='' + $F(''query_tags'')", :loading => "new Effect.Appear(''comment_loading'')", :complete

Hi there, I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also. I can't use IAX to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>|

Anyone have practical experience using inexpensive GAP-compliant DECT handsets with the Snom M3 basestation? When I asked Snom support, the answer was that 'basic functionality should work', but they didn't elaborate. I'm _guessing_ that means registering/unregistering with the base, making calls, and receiving calls (including presenting caller ID). They also stated that they

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten => s,1,Wait(30) exten => s,n,Answer ... exten => s,n,Dial(SIP/brian&SIP/joe,10,H) exten => s,n,Voicemail(u2001) exten => s,n,Hangup exten => s,103,Voicemail(u2001) exten =>

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user

Hi I have been playing with * for the last couple of weeks now. I am also speaking to one of my customers about installing a * server in addition to their Philips IS3090 switch. They are busy building a new office block and I have convinced them to go VoIP. Currently the client is thinking about a Philips IS2000 Voip switch. I am thinking * solution. 1) Would I use one the T1 cards

Hi, I've been very puzzled lately. I installed a phone system for a friend a few weeks ago, and they're having a problem that I can't get rid of, actually 2 problems. Before I go into the problems, let me tell you about the setup. It's a pretty small setup with only 4 handsets, all Polycom 320s, the server is a Dell SC440 with Intel E2180 CPU (dual core, 2GHz) and 512MB Ram.

Hi! Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone directories, and if so, how? Thanks in advance! On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >