similar to: Incorrect CDR results

I ripped a rock-and-roll CD for a client's moh. But it's too loud. Is there a simple way to reduce the gain without having to remix the tracks? Thanks -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0

Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.

I'm seeing quite a few "Not Found" pages when I google lists.digium.com. Is anyone else getting this? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

The attendant attempts an attended call transfer (all phones are IP501). The attendant pushes "hold", "transfer", dials the extension and announces the call. When the attendant pushes "transfer" the second time, the original call is lost. The reason this is a big problem is that the PRI channel for the call remains busy. Subsequent inbound calls on that

Is there a programmatic to to trim the silence from the beginning and end of a recording? From a .wav file? From a .ulaw file? Thanks, -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 mike@TelecomMatters.net www.TelecomMatters.net

I would like to capture the lat/lon coordinates from a GPS-enabled cell phone or PDA. Is this possible? Must I subscribe to this information from the cellphone network provider, or can I capture it without charge? What devices will broadcast the coordinates? Is there a device that will broadcast its position inband that can be captured by Asterisk? Can an SMS message include coordinates?

I need to test QoS on an IAX link between a server in Colorado and a server in Europe. I know I could install a Milliwatt extension on the European server and just listen, but is there a more scientific method to collect QoS metrics? Thanks P.S. I'm getting a lot of "Page Not Found" on lists.digium.com. Are the older posts being purged? -- Michael Welter Telecom Matters

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (even breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

I'm experiencing a very strange problem with SIP calls with a CLEC (CBeyond). The downstream audio with the telephone on mute is excellent. However, when there is upstream audio (i.e., breathing) from the mic, the downstream audio is clipped and sometimes dropped. The strange thing is, if I Monitor the call, the downstream audio in the wav file is perfect, even though there was clipping

Can someone help me, im using latest CVS, asterisk and cdr_mysql, when I make a bridge call (using .call files in outgoing/) I always get 'billsec=12' in the cdr, both mysql and Master file even if the call lasted longer, watching the Master file while making a call I see it updated at 12 seconds even while im still 'in' the DIAL app and the call continues on just fine. Iv looked

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Hello, I have been getting outbound nufone calls dropped after about 70 seconds. CLI shows "Attempting native bridge of IAX2". I have put "notransfer=yes" in iax.conf in the main section and all identifier sections. I tried a Tt in the dialstring, but it still tries the bridge. a cvs update didn't help. internal *server1 tdm fxs pci card <--> iax2 trunk -

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

Hi people, when I try to get the billsec in the dialplan, it is 0... but if after that I check the database, it is right (not 0). I'm trying to get it in the h extension, like: exten => h,1,Noop(End) exten => h,n,Noop(Time is ${CDR(billsec)}) .... Is it updated after the extension h is executed? In that case, how can I get the call duration in the h extension? Thanks, Anahi

Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the

Hello Asterisk-Users, For some reason my CDR records for disposition and billsec are not working correctly. I always receive a 0 for billsec and the disposition is always at "NO ANSWER', even when I grab the calls. I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22. Here is information on how I do the call: -----------------------------------------------------------------