similar to: IAX voice distortion with full upload channel / SIP ok

On Samstag, 14. Januar 2006 1:47 tim panton wrote: > That is weird, you would expect IAX to do better than SIP (bandwidth > wise) My point exactly. > 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. > 2) what codecs are you using. Are the codecs the same for IAX as > for sip? G.711 alaw and

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

Hi All, At present calls over IAX2 (ilbc) are good but they suffer from occasional distortion. The strange thing is that the distortion can only be heard by the calling party and not the called party in 95% of cases. IAX2 is being used with trunking enabled, using the ztdummy module as a timing source. Bandwidth shouldn't be an issue as there is more than sufficient plus we use QoS

hi All, Our configuration is, ISDN PRI lines connected to Asterisk Server, SIP users connected to Asterisk We route calls from ISDN PRI to SIP users, We did a CVS upgrade few days ago, now the sip users(CISCO phones) are experiencing, distortion of voice while they are engaged in a call. ie, SIP users can here the call clearly, but the outside caller heres distorted voice. ie, to

Hello, After checking out CVS HEAD from yesterday (for those new PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom IP600's. After seing it resolved as of this morning (thanks Mark), I decided to try again... I can answer incoming calls. No problem there. Putting calls on hold, however, results in my Polycom IP600 indicating the call on hold, but the caller does

I am having almost the exact same problem. I have the following setup: Debian Woody Kernel 2.4.18 CPU: P4 1.2GHz RAM: 1GB Asterisk - Latest CVS 1 TDM400P card Codec GSM I've been chasing down bandwidth issues, but have had no luck. We are still pursuing those issues. I just started configuring IAX, so I assumed it was related to my IAX configs. We just noticed this morning that SIP is

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

Hi, We have Asterisk, running on a P4 box running Suse 9.1, making calls using IAX through SimpleTelecom and Nufone. What we are looking for is toll quality voice. The problem is that voice over calls routed through SimpleTelecom and nNufone occassionally breaks. We also have a digium card and the calls over the digium card using the Zaptel Interface have a very good quality. We

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

Hi to all in a scenario where: - the bandwith is shared with other traffic (HTTP,VPN,ecc) - the PBX is on a remote VPN peer - due to many reasons Qos is not usable There is a IAX trunk between 2 Asterisk 1.4 i've tried different codecs (ulaw,alaw,gsm) but the main problem still remain the same: too many voice packet get lost. The main problem is surely on the network, but the strange thing

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

Anyone have an example of using SLA with SIP handsets in Asterisk 1.6? I'm looking at the sla.tex file and wondering when it was last updated... Thanks, -Philip

I've been having random problems when I make long distance calls using either VoicePulse or Nufone. Sometimes the calls go through clear, and other calls (or even just part of a call) the person on the other end just hears garbled voice, or really broken up voice. Sometimes it lasts for only a few seconds, but other times it goes on for a few minutes until I give up on the call. At