similar to: Rewind MusicOnHold?

> Hello > as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. > Only at restart the new files are available. > Is this a bug ?? How can I get more debugging for this problem ?? I think there is currently a bug with MOH. For now, if you add a file to a moh folder, 'touch musiconhold.conf' and then reload moh.

I have a question in that which is the best WINE to use? Or what are the differences wineX - This is the wine from Transgaming that the CVS does not contain SafeDisc but the purchasable one does? rewind - This is the MIT/X licensed version wine - This is the LGPL version from www.winehq.com Right now I use "wine" and update from CVS from winehq.com. So which is the best one to use as

Le 23/03/2017 ? 20:17, Jonas Kellens a ?crit : > Hello > > > is there any more information on how to reload/read musiconhold files ? CLI> module reload res_musiconhold -- Daniel > On 07-03-17 10:46, Jonas Kellens wrote: >> Hello >> >> I did not mention it but of course the MOH directory is listed in >> /etc/asterisk/musiconhold.conf : >> >>

I use IP Kall to forward my missed cell phone calls to. This way, if my phone is off, or out of a service area, calls will go to my * box. Concurrently, all incoming calls to my * box cause it to dial my local extensions at home, my extension at work, and my cell phone via NuFone. Problem: A loop can be created if my cell phone is not on. Say a call comes into my * box, it uses NuFone to call my

Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com

Has anyone had problems getting their preffered codecs on the Teliax web interface taking effect? I have two accounts, two separate yet similarly configured * servers. On one account the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the old codecs are usable (otherwise * says it can't negotiate a codec). Teliax

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for

I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

Dearest List, I guess I missed this point: Is it true that if you change the echo canceler in zconfig.h, and then recompile/install your zap modules, that for this to be taken into effect by * you must then recompile/install *? I would have figured that the zap echo cancellation method was independent of *, and I don't recall seeing any docs mentioning either way. Sincerely, Brent A.

Is it possible to dial more than one extension with a different CID to each extension? I'm thinking macros might be needed, but I don't have a good handle on macros. Is it possible? Any hints? BTW - this would be used for showing an internal extension to one phone and a PSTN accessible number to another phone. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road

Howdy: Does echo only occur on analogue PSTN lines, or can it also occur on PRI and BRI lines? If so, for the same reasons? This is a part of our consideration to transition to BRI. Sincerely, Brent A. Torrenga brent.torrenga@torrenga.com Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Hi, I have got my setup almost how I would like it now, but I have just two last remaining issues that I cant seem to find answers too so i'd be grateful if someone could help? 1) Since upgrading my Cisco 7960 SIP phone to P0S3-08-2-00 the phone now displays the IP address of my asterisk server alongside the caller ID of the incoming call. For example "0123456789@192.168.0.1",

I am thinking of turning on the NAT option in our Cisco phones (and the corresponding sip.conf modification) to allow the phones to be taken outside the LAN. Can anyone think of any reason not to just always turn on the NAT enabled option? I can't think of a reason not to always operate these phones with this enabled, since it would likely allow them to be taken outside our LAN and used.

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

I think A@Home allows a user to search a directory by either first OR last name, right? I don't know for sure since I don't run A@Home. I would like to offer that functionality in my system - and I'd have done it by now if there was a prompt where Allison asks "press 1 to search by first name, press 2 to search by last name". But I don't think that prompt exists. Can

Does anyone know where i can find out how powerful a machine has to be to handle a certain amount of call volume? Eg, 2Ghz is enough processing power to maintain 100 calls at a time. 4Ghz is engouh to process 250 calls etc etc. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]