similar to: iax connection and 1 way distortion

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

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 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, 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

Hi All, I've run into a small hiccup in encoding my audios with Speex. When I encode audience laughter and applause with 'speexenc' (version 1.0.1), the result is quite acceptable... until I enable VBR. Then it distorts horribly. My understanding of VBR is that it frees the encoder to vary the number of bits emitted to better maintain the quality requested, and so I would have

Hi all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This

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

Hi everyone, How can I introduce some distortion, echo, chopping sound and all other bad quality things that can happen to a SIP trunk? I have plenty of bandwidth and crisp clear lines so the only thing that I can think of is to limit bandwidth but even that requires quite some scripting work. Is there any easy way to simulate a distorted SIP line temporarily for testing? I am appreciate

Ok maybe it's just my imagination, but it looks like the one-line "duh" fix I committed from Andrej Vakrcka <ander@cauldron.sk> this week DRAMATICALLY reduces distortion in encoded files. Take a look and compair. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

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 =>

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

I'm a new Asterisk user. I've managed to set it up to do everything I want except sound good. Currently, Asterisk sounds considerably worse than my cell phone. I know VOIP can be _better_ than my cell phone, because I've heard Skype do it. (Using 32k iLBC, I believe.) I did an experiment with audio quality: 1) I made a recording which was pretty good. I used an iSight

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

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

One of my user is praising Skype!!! I cannot figure out anymore what I can improve! This users sip show peers is jumping from 65 msec to 1800 all the time. Of course his voice quality is like a morse code with dashes or dots of connection time. The next minute he calls me via Skype and it works fine !!!! What indicates that there is no fault on his Internet connection!!! He is using his

This is a newly installed server running CentOS 4 with all current updates. It's been cruising nicely since installation a week ago until this morning. Then everyting started timing out and the attached messages appeared in the log (this is a short and sanitized extract). Load went sky high. Had to reboot to restore service and all looks happy again now, but if anyone can provide some

Hi there again, I just wanna tell you what little bug we've found... Namespace is in mbox format. After deleting an mbox the entry in the .imap stays so not everyting is moved away and ist just hanging around... not so bad, but would be nice if it was cleaned up... thanks Tobi -- Dr. Nagler & Company GmbH Hauptstra?e 9 92253 Schnaittenbach Tel : 09622-7197-38 Fax : 09622-7197-50

Hello, I am creating a number of objects that I wish to have a common name with an index such as x1, x2, x3, ... I would like to do everyting in a loop to make the code compact and minimize the probability of an error by typo. A test problem may look like for (j in 1:10){ as.symbol(paste("x",j,sep="")) <- j; } which ideally would produce x1 = 1, ... x10 = 10. However,

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

I just installed Dovecot on Debian and everyting seems to be working fine except for the fact that it can't seem to find my mails when I log in. All my mails are stored in the file ~/mail/Inbox. In the dovecot.conf file I put: default_mail_env = mbox:~/mail/:INBOX=/var/mail/%u and it's not working. What is the part: mbox:~/mail/ for ? and what is the part: INBOX=/var/mail/%u for ? I