similar to: Packet Truncated - Choppy Audio

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) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

I installed ntop on a centos5.2 system (which got all yum updates a few days ago)... # rpm -q ntop rrdtool ntop-3.3.8-1.el5.rf rrdtool-1.2.28-1.el5.rf went through the password configuration when I start it, its erroring. # service ntop start Starting ntop: Processing file /etc/ntop.conf for parameters... Mon Nov 24 23:56:36 2008 NOTE: Interface merge enabled by default Mon

Hello all. I have a strange and irritating problem with Star Wars: Knights of the Old Republic. I played ok up to a place where I saved. There, I quit the game. When I restarted, the sound became choppy, making the game unplayable: about 1/2 sec of sound plays, then loops for about 3-4 sec, then the next 1/2 sec, loops again, etc... The screen isn't updated during this; well, it is updated,

I don't know why I haven't signed up for this list before since we use CentOS all over the place. The list is very useful and it is good for me to participate and "give back" to the community. Anywho, I wanted to post this response to a thread that was created back in November 2008 about the ntop daemon failing to start. I'm currently setting up ntop as a NetFlow &

This email is intended to document an issue for anyone searching the archives. We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable conversation could be established due to extremely choppy audio in one direction only (outbound from * to distant sip phones and distant * boxes). We were running HEAD from June 8th. While diagnosing the root cause, we monitored bandwidth

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 better off using sip with them? Is there

Hi, I'm working on some project which is sort of log filter. Last few days I noticed that there are some wacky people scanning sshd port all the time from anywhere. Although sshd reports it with syslog error message which is very helpful, I'd like to know the source ip address with following message: canohost.c: around line #100 if (getaddrinfo(name, NULL, &hints, &aitop) !=

Hi, Can anyone please recomment an ntop alternative for me, which is more stable as well? I need to monitor all connections to and from a CentOS 5 server and ntop does it fairly well, but seems to crash at random times, and thus looses all the date prior to the crash. Cacti / MRTG only gives complete bandwidth usage on the given interface, but I need to know how much bandwidth goes where, and

Hi SSH developers, I am wondering if someone could help explain a behavior of sshd. From canohost.c, get_remote_hostname(), it seems sshd will always try to reverse lookup the ip address of any client that attaches to it. The verify_reverse_mapping flag just turns off the forward lookup through DNS of the clients hostname, once the hostname has been determined. I am using Solaris 8 with ssh

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

Hello all! If I place a call to our number, the call is routed to our Asterisk box from teliax --> IAX2 --> firewall w/ port forwarding --> * If that caller dials an extension that rings an outside line, where our * box makes an outbound connection to teliax to terminate the call, we get choppy audio. Internal extensions have been dialing outbound calls no problem for over a week. What

If you are running ntop on 5.4, what compile options? Use ports version? Or surgefile tarball? It makes a great security forensics tools, but I can't get it to stop segfaulting.Was wondering if anyone found a fix for it. -- Michael Scheidell, CTO 561-999-5000, ext 1131 SECNAP Network Security Corporation Take a vacation from spam: http://www.spammertrap.com

I'm changing some functions from storing data in SQLite (using RSQLite) to storing it in PostgreSQL (using RODBC). When trying to store very long character fields I get the following message: > sqlSave(pg, Grids, rownames = FALSE, append = TRUE) Warning messages: 1: In odbcUpdate(channel, query, mydata, paramdata, test = test, verbose = verbose, : character data truncated in column

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

If bind() fails we _always_ should close socket. I sent this patch while ago to djm but I still don't see this fix in openssh_cvs. diff -urN openssh-2.3.0p1.org/sshd.c openssh-2.3.0p1/sshd.c --- openssh-2.3.0p1.org/sshd.c Sat Jan 6 19:54:11 2001 +++ openssh-2.3.0p1/sshd.c Sat Jan 6 19:55:48 2001 @@ -782,10 +782,10 @@ debug("Bind to port %s on %s.", strport, ntop); /*

Hi all, 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

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 Guys I'm conecting 2 astersk 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

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

Dear List, Sorry for the dublicated email but I couldn''t get any answer. I am trying to limit some IP blocs with tc with following three step. # interface tc qdisc add dev eth0 root handle 1: cbq avpkt 1000 bandwidth 256kbit # class tc class add dev eth0 parent 1: classid 1:1 cbq rate 64kbit \ allot 1500 prio 5 bounded isolated # rules # download tc filter add dev eth0 parent 1: