similar to: Supported: ms-early-media

The Wine development release 1.3.23 is now available. What's new in this release (see below for details): - Support for stubless COM proxies on x86-64. - Builtin dxdiag now outputs real information. - Monochrome bitmap format in the DIB engine. - Beginnings of a true shell Explorer builtin. - A number of new D3DX9 functions. - More support for Indic text shaping. - Various bug

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

Greetings, upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when the PSTN party answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and

Hi all: I was analysising the speex code these days and learned much from it. Thank you so much! But I have a question about VAD: After preprocess(denoise and agc are off, vad is on),the input pcm speech was distorted seriously.The input speech include ten syllables(one~ ten), but the output speech lost one! Can you tell me why? Please see the attachment about input and output: (See attached

I am working with a largish dataset of 25k lines and I am now tying to use predict. pred = predict(cuDataGlmModel, length + meanPitch + minimumPitch + maximumPitch + meanF1 + meanF2 + meanF3 + meanF4 + meanF5 + ratioF1ToF2 + rationF3ToF1 + jitter + shimmer + percentUnvoicedFrames + numberOfVoiceBreaks + percentOfVoiceBreaks + meanIntensity + minimumIntensity + maximumIntensity +

I am on Ubuntu 9.10, and I am trying to get Anarchy Online to work. Wine with it works well on my other computer, which is Arch Linux, and much more modern (actually has graphics card). My current computer has 1 gb ram, 3ghz processor. This is what happens: Code: *****@*********:~/Desktop/Anarchy Online$ wine Anarchy.exe fixme:actctx:parse_depend_manifests Could not find dependent assembly

Hey there! I've been playing Evil Genius on steam and for the most part it works fine, but frequently (sometimes more frequently than others) it crashes at very inopportune times. The wine page for the game talks about unexpected crashes, but it sounds like these are related to the game autosaving, and I turned off my autosaving yet this persists. Here's what my terminal (The first bit

Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols

We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk 1.6 and theirs is 1.8. I tried promiscredir=yes and then the calls fail altogether because rather

hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone & POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone -> asterisk -> PRI -> phone co. i call the same cell# and if it's unavailable. the PRI return

I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send

Hi,all Does asterisk support sip early media? I have a setup asterisk for sip ATA boxs and a SIP trunk (SIP GATEWAY) for PSTN access. The ATA can call PSTN phone, cell phone, BUT it can't receive early media. I am sure the SIP GATEWAY support early media. If use the ATA connect to the gateway directly, it can receive early media. Jiangzhou Best Regards -------------- next

Hi, I've searched almost everywhere but have not come across a solution so I was hoping one of your fine folks can help me out. The problem is that a carrier is passing me early media on calls that sometimes have problems connecting. For example, calls to India mobile might play an early media message saying "the phone is out of reach" if mobile is out of area of coverage. Problem

Hi using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call it should goto my specified extension. my php script: $oSocket =

Hello! I am trying to come up with a way to read a digit *before* the call is answered. My Asterisk version is 1.6.2.0-rc6 SIP early media works fine (I can receive and transmit audio before the call is answered), but as soon as I start the read application, Asterisk answers the call which is not what I want. Here is how to reproduce the problem: send incoming calls from a SIP provider that

My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was using Dial() to call all of my dahdi (TDM400P) extensions. The results were that incoming calls would not hear any ringing tones and the call would be ended by Teliax after 21 seconds. Looking at the packet dumps,

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer Can someone clarify what "early media" is? I noticed that NOT answering a call before dumping them into a queue that has music on hold will not set up a leg to push music back over the calling SIP channel. Tossing an Answer command into the dialplan just before moving to the queue alleviates this (in either situation the

I have a dialplan (freepbx) that plays a busy signal in-band when an extension is busy (before an Answer). Stripped down, it looks like this: exten => 1005,n,PlayTones(busy) exten => 1005,n,Busy(20) Note that there is no Answer() prior to this. Our trunk is a PRI. When I call into this extension from outside, I get about 25 seconds of ringing, followed by a hangup. Looking at the