similar to: Sound Distortion using IAX?

Displaying 20 results from an estimated 10000 matches similar to: "Sound Distortion using IAX?"

2006 Jan 14
2
IAX voice distortion with full upload channel / SIP ok
Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote
2005 Oct 07
1
Distorted VM with iax2 with ilbc and jitterbuffer - bug?
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
2004 Sep 29
1
iax connection and 1 way distortion
I'm new to * and I have my * box connected to another * box located at my ISP via iax2. Using ilbc everyting works fine but with any other codec I've tried (gsm, ulaw, alaw) there is severe distortion on the sound going out of my box as well as a second or two of delay. The incoming sound quality is fine. This happens even if the outgoing sound is coming from the voice mail prompts on
2005 May 22
1
Upgrade cause's no Audio on IAX
Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction.
2006 Mar 21
1
Problem with chan_iax.c implimentation causesbadaudio?
We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the
2006 Jan 16
1
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
2004 Apr 07
1
Out of trunk data space on call number 16386, dropping
Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes
2006 Jan 19
1
Sound issue with Asterisk
Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am
2004 Dec 12
1
Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work. I will in the end only have 4 SIP extensions being either softphones of IP phones. Currently only 1 SIP config for testing. And at the this point it should be all fairly easy with all inbound and outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via IAX. Inbound does work in it's current basic state.
2006 Mar 21
1
Problem with chan_iax.c implimentation causesbad audio?
We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried
2005 Feb 12
1
iax.conf config and iax based clients
Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect
2004 May 24
3
Help with IAX , voice Distortion or Breakage.
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
2003 Nov 12
1
IAX needs a zaptel device?
Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to
2004 Dec 22
0
Ticket: 12775 Multiple IAX client behind a NAT
Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1
2010 Feb 08
0
Help with iax.conf {tesco|freshtel} 1.6
I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382
2006 Feb 20
9
Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
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.
2006 Jun 28
2
Standard Sound Files Distortion
I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. I did a little test. This sounds fine... exten => 1000,1,Answer exten => 1000,n,Wait,1 exten =>
2007 Nov 20
1
Switch to Multi-Proc -> Choppy sound?
Hello, everyone I'm relatively new to Asterisk (and VOIP in general), but I have a project that it will really help with. So, I setup a test system on an ancient 400MHz P3 we had lying around. It worked great. I had a test dialplan working, and had no trouble connecting to it with SIP using 3CX SoftPhone over our LAN (and over the Net through our NAT). So, we went ahead and bought a
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes
2005 May 13
0
Problem with IAX trunking
Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following