[Asterisk-Users] Re: Who's happy with their voip service?
I started out happy as a clam with my new Broadvoice account and asterisk machine. About 10 days ago things began to change Who's happy with their voip service using asterisk? Where do you get reliable DIDs? The 'carrier partner' they speak of.. can you get the did directly from them? Are all the voip providers this flakey? I've tried 5 providers, and I can't say that I'm happy with any of them. I'm far to small to deal with 'carrier partners' directly (e.g. Level 3, XO or RNK). So I have to deal through resellers. And they all seem to be operating on shoestrings and duct tape. I'm OK with the awkward setup, confusing configuration, and (for Asterisk) all but useless documentation. But high latency, dropouts, unplanned outages, lack of clues, echos, all take the shine off things. With Asterisk I have very few tools to monitor connection quality, especially on the outbound leg of my calls. I at least want to know when it sucks, and have some control over parameters. I keep a POTS line at home. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting the jitter buffer in AIX
Are these things possible? 1) Set the local Asterisk jitterbuffer size, but only for a particular connection. I'd like to force Asterisk to use a particularly large buffer in certain cases. Should I expect this to work? [general] jitterbuffer=no register => username:[EMAIL PROTECTED] ;parcelfarce register => username:[EMAIL PROTECTED] ;iaxtel [parcelfarce] ;connection to parcelfarce type=friend auth=md5 secret=password context=inbound-from-parcelfarce host=parcelface.domain.net qualify=yes jitterbuffer=yes maxjitterbuffer=600 2) Set the remote jitterbuffer. I want to tell the remote Asterisk that, during this call or part of a call, that a much larger jitter buffer is OK. Basically I care more about quality of the delivered sounds, rather than latency. 3) Monitor the remote jitter buffer discards. I want to know if my outgoing stream is breaking up. Here I am building an information retrival service, so conversational latency is not an issue. The remote Asterisk is at a PSTN company such as VoicePulse, TelAIX or LiveVOIP. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Optimal prompt format (gsm, ulaw, wav) for quality effeciency space
I'll summarize on http://www.voip-info.org/ What's the best way to store static prompts in Asterisk? The system prompts are provided in gsm format. But we have no limits on disk space. Is there a better format? If we provide clips at 8 bit 8000Hz mono ulaw, and our IAX connection matches, do we gain in speed, efficiency, and quality? Will Asterisk search for the best matching clip, if we provide several formats? Do I have the sequence right, for a the media stream of a voice clip? GSM format on disk. converted on the fly to uLAW by our Asterisk box Transmitted via IAX to a (Broadvoice, LiveVOIP, VoicePulse) data center converted on the fly to Transmitted via SIP to a (Level 3, RNK, XO) data center Transmitted to a point of presence converted on the fly to TDM Transmitted via the PSTN converted on the fly to analog Transmitted via last mile copper converted to sound waves by a telephone converted to brain waves by an ear Some of our prerecorded clips sound OK at the end. The clips generated by Festival Text-To-Speech sound like an Edison wax cylinder was used an an intermediate codec. Try 970-688-4348 and enter code 511#. Obviously we'd like high quality, and lack of dropped voice packets. In our particular case latency is not an issue at all, the user is simply hearing a mesage. -Bryce Nesbitt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
