similar to: Bad audio quality on SIP

Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)?

Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/96b14a3e/attachment.htm

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

Hi again, folks. I've been getting feedback from this list and elsewhere that softphones are generally not considered good enough for hardcore business use. Can someone point me to where I can find more detail on this debate? Is the problem that the technology isn't mature, that the load on the computer is too high, or simply that it doesn't work well in a poorly designed

Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in sip.conf but not iax.conf Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091001/abf55045/attachment.htm

We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB

Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites) When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is <175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

I found some great pieces of script on the internet that I've combined to allow Asterisk to send voicemails as an MP3 file, and encode the sender name and number as well as message number as tags into the MP3 file. I even include a "cover art" image which has our company logo and PBX symbol in it. Mobile phone users love it, and Android phones can now play the attachments (without

I have a new asterisk system with a T1 card. It appears that running "ztcfg -vv " is required in order for asterisk to start properly. Is this correct? Are people adding this command to the asterisk startup script? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

We're creating a SIP gateway for a client that will take one leg of a call in via SIP, and out the other side via H.323. To minimize load on the gateway, we would like to have the RTP stream bypass the gatewayy altogether (directrtp/reinvite). Is this possible with these to protocols? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks!

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm