similar to: SIP devices with packet loss tolerance

We are evaluating starting a small VoIP consumer based platform. What is the best codec to use with customers using primarily DSL as internet connectivity? I know that g729 is the king-all, but I want to know what the rest of the professional are using out there. g729 has a cost involved, so does the cost really offset the performance? Or is it better to go with g711 to start off? We plan

Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Greetings list, Some time ago (probably about a year ago now) we disabled IAX jitter buffering on all our boxes because it was causing issues in a mixed 1.0 and 1.2 environment. One thing I've noticed over the last few months as more and more clients have moved from the 512k/1mb/2mb ADSL connections they were using onto "up to 8mb" connections is that whilst overall throughput is a

Hi Ross, thanks for this. It appears there was some problem when asterisk went from 1.07 to 1.2 The non-commercial codec providers are aware of this but dont know when/if they will be able to fix this (this obviously also affects anyone who is running asterisk@home 2.0 and up) Guess if it's a big enough problem buy a commercial codec etc. Cheers, Dean -----Original Message----- From:

Greetings all, I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. I know there are lots of Speex

Greetings list, I'm coming across an issue with some of the GXP-2000 phones we have out in the wild at clients' employees' homes. In most cases they're behind consumer ADSL NAT routers on a dynamic IP from their ISP. In a nutshell, the phone is unable to be called unless it's restarted first, after which it's fine for a good few hours, then it stops working until

Hi, does anyone have test numbers in Germany that would allow me to tune my rxgain/txgain settings? I know there are numbers provided by other providers in UK e.g. but have yet failed to find a number in Germany (esp. by Deutsche Telekom). Kind regards, JP

Greetings list, Quite a few of our users seem to be experiencing poor voice quality when they're using internet connections over which we have little or no control (i.e. they're using their own router with no QoS, etc.). Some of these connections are giving a qualify time within asterisk of 130ms+. Are there any recommendations as to phones with particularly good buffering that might

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it

Greetings list, I've noticed over the last couple of weeks that, unsurprisingly, nearly every new PC seems to be coming with Vista these days. I expect it'll only be a matter of time for all of us before clients start needing Vista-compatible softphones (if it's not already happened). So, what's the story with Vista compatibility amongst the softphones currently out there?

I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral

Greetings list, I remember a discussion many months ago which ISTR concluded that asterisk didn't play nicely at all in multi-homed setups (e.g. SIP packets not being sent out through the same interface they were received on, etc.). Is this still the case, or are there versions which have resolved the issue? Even if it's still the case, is this only a problem for SIP, or does it affect

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL

Hi, Is it possible to dial voicemail from a particular phone line and automatically enter the extension that is being dialed from, thereby only prompting for the password? I've been searching around to find if this is possible, but I haven't been able to find an example of this. I have a feeling it's more of a endpoint function, but I thought I'd ask if anyone has accomplished

Hi guys, Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? Alex

Hello, I'm experiencing some problems with running IAX2 protocol on quite reliable link with G729A codec. My customer has 2mb FR link to the Internet used in about 20%. Ping statistics: 50 packets transmitted, 49 received, 2% packet loss, time 49496ms rtt min/avg/max/mdev = 9.308/13.126/33.307/4.851 ms Everything would be great, but the quality isn't good enough. I have 2mb/512kb DSL

Hi, I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to make zaptel working... OS is gentoo linux 2006.1 Hardware: --------- 0000:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 22 I/O ports at b400 Memory at ff900000