similar to: IAX2 update - timestamp issue within iax pkts

I am wondering if there is a good way to work with data that is indexed in time, via timestamps with a resolution in milliseconds. As I understand it, the POSIX classes have a resolution i n terms of seconds, and will not process fractional seconds from a string. Is this correct. I realize that this may be a little unclear. Here is what I am trying to do: A data frame with a time series

Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to "register" like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm using canreinvite=no to forcably keep * in the middle for now. Questions: 1.

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

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

I have this script which I use to get an epoch with accuracy of 1 second (based on R's inability to calculate millisecond-accurate timestamps -- at least I have not seen a straightforward solution :) ): nowInSeconds <- as.numeric(Sys.time()) nowInMS <- nowInSeconds * 1000 print(nowInSeconds) print(as.character(nowInMS)) when running this I get the following: > nowInSeconds <-

I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from "certain" interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is

Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again ..... Alex

Hi all, Just hit this problem for the first time: WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from address 10.25x.xxx.160 When I ran "iax2 show peers" everything comes up as unreachable, no calls are passed between the servers (as would be expected) but there is no problem with DAHDI / SIP channels. At the same time there are no network issues (can ping all

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 thought the same thing before I made my reply but zapata.conf seems to be the only config file that deals with echo at all. From what I understand of 'echotrain' is that at the beginning of the call it sends a short signal out that measure echo in an attempt to try and cancel it. I was wondering if you tried using it and if so was it of any help? Sincerely, Steve But is Zapata.conf

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 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

Hi, apart from the mmap(2)ed DBM files that Samba uses, are they any cases where Samba will *modify* data files without setting the mtime ? I have issues with rsync not seeing changes to Samba exported files (md5sum don't match). The mtime is however in the very distant past (say 2004), but the content seems to have changed. I don't think it's a data integrity issue. Any idea ?

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls - very stuttery , have tried a few different codecs. I assume this is a granular clocking issue , and wondering if anyone has anything I could try to fix or mitigate the problem in ESXi environment . We have

Current dev cvs install on two systems. System A is behind a SonicWall firewall, and system B is on a registered IP address. (System B has multiple iax links that are fully functional to multiple locations.) System A is correctly registering with System B, with no special firewall rules. Should System B be able to take advantage of the "registration" to send iax/gsm calls to System A

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmware version:

Maybe this is the right place to ask? Thanks again, Daniel Begin forwarded message: > From: Daniel Schaffrath <daniel.schaffrath@mac.com> > Date: 21 July 2007 11:01:01 GMT+02:00 > To: linux-net@vger.kernel.org > Subject: PFIFO to contain more pkts than allowed by TCP peer? > > Hello everybody, > > when the one and only connection originating my box is a TCP stream

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect

Is there a easy way to create the time index for a zoo/xts object for every 100 milliseconds. eg. time Index would be: 10:00:00:100 10:00:00:200 10:00:00:300 10:00:00:400 I am looking to build an empty zoo/xts object with time index from 10am to 3pm, index jumps by 100ms each row. Thanks, Chris -- View this message in context:

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=488 Summary: Chain/Groupings of networks don't total pkts and bytes correctly Product: iptables Version: 1.2.11 Platform: i386 OS/Version: RedHat Linux Status: NEW Severity: normal Priority: P2 Component: iptables