search for: choppy

Hi folks, I'm experiencing some problems with sound through phpAGI ... What I'm trying to do is a menu, doing some database lookups and so ... But sometimes the sound become too choppy ... just sometimes .. like 1 of 5 calls ... but is a big percentage ... And I have my current menu on the dialplan that I have no problems with it ... I'm using .gsm for both but different recordings ... Does anybody has had problems like that ? Is it AGI performance problem ... even the aten...

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec)...

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be bette...

Hey all, I''ve got an interesting one for anyone who''s up for a challenge. Essentially, I have a very choppy effect, that almost looks like timeouts are overloaded or interfering or something, that only occurs when sortables are on the same page as "standard" effects. Here''s what I''m doing: I have a menu that slides in and out on the right side of the page. I also have a s...

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PS...

...I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data were lost. The strange thing is that the x-lite user hears the PSTN user fine ! In x-lite, I have swithed off sience detection (transmit silence - yes), this has improved the sound quality but did not eliminated the...

...ing test). Codec is alaw or ulaw. TDM card is plugged to an NEC PBX (old NEAX 2000 IVS) via the analog station card in the PBX. PROBLEM: Establish a call from the GS to an NEC phone (Dterm III) connected to the PBX. The voice quality on the GS sounds good. The voice quality on the NEC gets very choppy (random). A call from the same GS to an internal GS here (same latency, same IP path) is much better (some chop). SOLUTIONS tried to date: Tried a local network test, sound excellent. Tried a 'low latency vpn' test bed to reduce latency (is latency the issue?). Latency down to ~70ms rou...

Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw => ulaw is choppy, ulaw => alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit-

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, b...

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 and C7960 are using ulaw; Both are 100 meg on local LAN. Top suggests all processes running less then 1 or 2 percent...

I loaded Unreal Tournament in wine 1.3.37 and everything works that I tried but the video is choppy. The sound was not choppy but the video was. It did not matter if I was playing online or the bots on my computer the video was choppy. I tried raising and lowering the video resolutions and it did not seem to effect the problem for better or worse. I am using Ubuntu 64 bit 11.10. I am on an AM...

...n asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when asterisk starts saying the digits from the extension, the sound starts becoming very choppy. The voice after the digits is still choppy. Does anyone have a suggestion? The codec that asterisk is using with the softphone I am using is the GSM codec. Please advise, Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermai...

...k servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)<====iax2 trunk====>(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Thanks Yelson

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any g...

All, A bit of background: I work on a QA API on a network that is very choppy (a lot of network interrupts), and we use ssh to do a large part of this automation. This leads to some problems: ssh connections seem to be sensitive to network state, becoming unusable if the choppiness reaches a certain threshold, and either timing out or disconnecting if this happens. Anyways...

....42-kernel.mp3 Mario -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Webster, Andrew Sent: November 12, 2006 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: Choppy sound in voicemail usingAsterisk1.2.11 on CENTOS4 guest on vmware server > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Martin Joseph > Sent: Saturday, November 11, 2006 15:01 > To: asteri...

I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing?...

Hi All, Whenever a call comes in via the ISDN and somebody leaves a voicemail, the sound file recorded is very choppy. If I actually take the call, the sound is not choppy so it's obviously something to do with the Asterisk box itself having to do the recording. Perhaps the sound card drivers? I'm using the stock i810_audio (OSS) drivers on Fedora Core 1. If I call from a local VoIP client to the Asterisk...

...ary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our system, they hear an IVR. When the caller dials an ext (SIP phone), the ringing progress tone is choppy/distorted... However, the voice call itself sounds fine. Asterisk, the Cisco phone, and the call gateway are all configured to use rfc2833. From my research, asterisk generates progress tones out-of-band (I think) unless turned on. We don't have any problems with the progress tones when G.7...

Hi After a day of running asterisk, I got choppy sound when fw ip->pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091009/be8bbf2f/attachment.htm