similar to: Asterisk for transcoding

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in

Hi there, up till now I had this two-box setup in mind: * no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the

In case there are any Debian developers around here, I wanted to point out message <20020828154322.GA15114@chulak.naquadah.org> on the debian-devel list this morning. Another Debian developer is proposing to submit an mp3->vorbis transcoding program for inclusion in Debian. I have objected to this on the grounds that the resulting vorbis files will sound like crap, and I have also pointed

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if

As far as I can see, transcoding could be usefull for people who do not primarly care about quality but about filesizes. One could assume that such a user would have a collection of mp3's at 128kbps or higher bitrates, and uses an encoder like BladeEnc or Xing. He wants to take uses of ogg's supposed quality and transcode his 128-or-higher files into 96 or 112kbps oggs to save diskspace.

Figured I'd join the list since I saw someone else had just popped in looking for exactly the same thing I am (non-transcoding opus streamer for sending multiple .opus files to icecast2). It looks like ices2 would do exactly what I personally need, except for not having been updated to support opus yet. (I chatted on IRC once or twice with someone who it sounds like is interested in adding

Hi, I have a bunch of old radio programs (mystery/drama shows, not music) encoded at 32 kbit (and some 48kbit) mp3 (mono). I want to reencode them in ogg and make them available over gnutella. My question is this. What is the best quality level (-q) for transcoding them. I want to preserve quality, but I want to be sensitive to the many modem based gnutella users. I also want to to

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

I created some test samples and transcoded to FDK AAC and libopus at fairly low bitrates - I cannot recreate what bothered me about Opus & noisy music previously. It also seems I cannot tease ffmpeg into encoding FDK's AAC with VBR. As it stands, Opus clearly wins in this scenario.* Q: Is it possible to stream in variable bitrate? * ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Opus or AAC will give you comparable results at reasonable bitrates (~128k). Though, I would suggest finding a way to get more storage. You could upload to Backblaze B2 or AWS S3 for pennies, if your current host won't let you upgrade. On Sat, Apr 15, 2023 at 3:36?PM D.T. <ohnonot-github at posteo.de> wrote: > Situation: > > - remote virtual server with very little

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

Good afternoon, I'm trying to use the CLI command file convert on an Asterisk 1.4.26 server with a TC400B transcoding card. The transcoding card is working well for calls but I have some trouble converting sound files from alaw to g729. The command creates empty file as you can see below... CLI> file convert /var/lib/asterisk/sounds/fr/service_notactivated.alaw

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

Hi, Given that the Digium transcoding card has no external connections (AFAIK), it strikes me that it would suit a mini-PCI slot very well. Does such a beast exist, or is it likely to? Am I correct in assuming that this is a Digium-only product, and there is no OEM equivalent "generic" board out there that I could be investigating? It would be such a shame to waste a PCI slot that

Hi, I use samba for one year only, but with full satisfaction. Only with latest cvs tree I discovered some (maybe particular) problem. In mixed environment of win95oemII and NT4 workst (sp3 or sp4) we share some Paradox 7.0 database on linux server (Rh5.2 and 2.2.0-pre1 kernel). Paradox controls multiuser access to the data with three files: PDOXUSRS.NET (user names etc?), and two lock files,