[asterisk-users] snom 320's creating inappropriate conference calls
With asterisk 1.2.17 and snom 320 version 6.2.23, if a user places a call on hold, then places another call and hangs up, the two calls are conferenced together. What is causing this and how to I stop it? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom 320's creating inappropriate conference calls
On Sat, Jan 3, 2009 at 10:55 AM, Steve Howes st...@geekinter.net wrote: On 3 Jan 2009, at 15:50, Michael Boers wrote: With asterisk 1.2.17 and snom 320 version 6.2.23, if a user places a call on hold, then places another call and hangs up, the two calls are conferenced together. What is causing this and how to I stop it? Its called an attended transfer... Its meant to happen... I appreciate the attended transfer capabilities of asterisk, I just thought that is what the transfer button was for. I need to know how to allow a user to juggle two different calls on hold without accidentally connecting them together. On some phone systems the sequence: 1. Receive Call #1 and place on hold 2. Place Call #2 and hang up Does not connect Call #1 #2 together. Is there a different sequence I should use in this situation? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking calls and snom phones
When I try to park a call using a snom 320 phone, the phone disconnects before I hear the parking spot announced. Is there a way to avoid this? I have tried the following: Press button programmed as extension 700 Press Transfer, then button programmed as extension 700 Press Hold, then button programmed as extension 700 Press button programmed as park orbit 700 Press Transfer, then button programmed as park orbit 700 Press Hold, then button programmed as park orbit 700 In each case, the phone disconnects immediately. Some park the call successfully, but you have to guess as to where. I would be open to any approach either audio or visual for learning the park position. Parking works fine from my idefisk softphone. Asterisk 1.2.16 Thanks, Michael Boers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Both of the zaptel (1.2 and 1.4) drivers I tried were the latest. Are the older drivers working better? Does anyone have a configuration close to mine that is working? -- Michael Boers On 4/6/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Michael Boers wrote: I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ameike I am experiencing the same thing. I assumed that I just didn't have a fast enough CPU (2.4 Ghz Celeron Ghz, also tried on a 1.8 Ghz Pentium 4). I am using a T400P with an Adtran TA750 Channel Bank rather than the Digium analog cards. I'm not doing any VoIP on this system, strictly analog and I get echo on calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
In my case, the rx and tx gains are 0. I will try the 1.4 version for fxotune to see if that helps. Thanks for the suggestions! -- Michael Boers On 4/6/07, Stephen Bosch [EMAIL PROTECTED] wrote: Hi: Greg Siemon wrote: Same issue here as well. Running Centos 4.4 (x86_64) with all updates installed on a Pentium D 2.8 GHz, 1.5GB RAM and TDM800P with 1 x Quad FXS and 2 x FXO ports. I installed the new 64 bit HPEC module earlier in the week when it was released and have been unable to get it to work. I have the same symptoms as the OP and also noticed that there is much more line noise (soft static) than when HPEC is disabled. I sent a support request in several days ago and I have not yet seen a reply. I have observed very similar behaviour in an HPEC installation. Some observations: the clipping happens if receive gain is cranked up. If I set the gain to defaults, the clipping goes away, but then I have the problem of users complaining that the remote caller is too quiet. Another recommendation -- whether or not you are using Zaptel 1.2.x or 1.4.x, you should always the fxotune from 1.4.x. A question - do the people who are experiencing this problem have non-default gain settings, either in zapata.conf or in the telephone sets? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Balancing the Hybrid
In an article on voip-info.org ( http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it is suggested that one thing you can try to reduce echo is to insert a 500 ohm resister on the ring line of the POTS line. Has anyone successfully tried this? What was your experience. Also, what size 500 ohm resister would you use? I am not an electrical engineer, but I can wire a circuit in series. Any other echo reducing suggestions would be greatly appreciated! -- Michael Boers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Balancing the Hybrid
Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I am in the US, so opermode should not be ok at default settings. I just recently got fxotune working on my system. The version that comes with zaptel 1.2.16 would simply hang. I am using the 1.4 fxotune now with the 1.2.16 driver. That has reduced the echo coefficient from 35% to 8%. We will see how that does. -- Michael Boers On 4/6/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Apr 06, 2007 at 04:11:45PM -0400, Michael Boers wrote: In an article on voip-info.org ( http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it is suggested that one thing you can try to reduce echo is to insert a 500 ohm resister on the ring line of the POTS line. Has anyone successfully tried this? What was your experience. Also, what size 500 ohm resister would you use? I am not an electrical engineer, but I can wire a circuit in series. Which adapter do you have? for most of those that are not x100p, start with setting a proper opermode. If that is not good enough, try fxotune. Both of these play with characteristics of the card that are equivalent to adding an extra resitor and also other similar changes. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Balancing the Hybrid
Oops, meant to say that opermode should BE ok at default settings. On 4/6/07, Michael Boers [EMAIL PROTECTED] wrote: Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I am in the US, so opermode should not be ok at default settings. I just recently got fxotune working on my system. The version that comes with zaptel 1.2.16 would simply hang. I am using the 1.4 fxotune now with the 1.2.16 driver. That has reduced the echo coefficient from 35% to 8%. We will see how that does. -- Michael Boers On 4/6/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Apr 06, 2007 at 04:11:45PM -0400, Michael Boers wrote: In an article on voip-info.org ( http://www.voip-info.org/wiki/view/Asterisk+setup+research+lab ) , it is suggested that one thing you can try to reduce echo is to insert a 500 ohm resister on the ring line of the POTS line. Has anyone successfully tried this? What was your experience. Also, what size 500 ohm resister would you use? I am not an electrical engineer, but I can wire a circuit in series. Which adapter do you have? for most of those that are not x100p, start with setting a proper opermode. If that is not good enough, try fxotune. Both of these play with characteristics of the card that are equivalent to adding an extra resitor and also other similar changes. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HPEC audio clipping
I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ameike If the taps are set to less than 16, there is no clipping, but there is significant echo. I don't know if this is relevant, but asterisk does report No Zaptel transcoder support on startup. I am at my wits end; any advice would be greatly appreciated. My setup follows: Hardware, AMD Athlon(tm) 64 Processor 3000+, 512MB ram, 2 TDM400p with a total of 5 fxo channels, snom 320 phones OS: gentoo 2006.1 amd64 Software: Asterisk 1.2.17 and Zaptel 1.2.16 (fxotune does not seem to work in this configuration) Also tried Asterisk 1.4.2 and Zaptel 1.4.1 (fxotune does work, but doesn't seem to help in this configuration) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended Transfer with snom phones
I have setup an asterisk based phone system using snom-320 (SIP based) phones. I would like to change what seems to be the default procedure for an attended call transfer. Right now, the phone user places the call on hold, calls the extension using a extension button on the phone, speaks with the call recipient , and presses transfer to transfer the held call. The users would like to press transfer, call the extension using a extension button on the phone, speak to the recipient, then hangup to complete the call. Can you give me a suggestion as to how to do this. Thank you, Michael Boers Michael Scott Technology ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
