similar to: bandwidth cosume - iax

I have this in my rsync.conf [backup] path = /home/$USER/backup use chroot = no monge symlinks = no uid = wendell gid = wendell read only = no list = yes auth users = wendell secrets file = /etc/rsyncd.secrets But rsync doesn't seem to interpret the $USER. I've also tried this with $HOME and '~' with no luck. I want to be able to back up to the

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

http://bugzilla.netfilter.org/show_bug.cgi?id=768 Summary: Returns the position the entry was inserted Product: iptables Version: unspecified Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P5 Component: libiptc AssignedTo: netfilter-buglog at lists.netfilter.org

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for

Howdy all... I am trying to use two different samba servers in a test environment such that a Win98 SE user logs into his/her workstation, authenticates/authorizes themself via the PDC, and then mounts a different samba server to store his/her files. By files I *guess* I mean profile as this is where I assume that this user's personal files will end up. Question 1: Is this assumption that a

I place a call from an IAXY to an IAXY device. INitially the calls show in the output of "show channels". Then after a few seconds the "show channels" command shows 0 active channels even though I am still talking on the channels. Any ideas on this? THanks, Jerry

Also sprach Michael Smith: > > >Ok. I'm using shout and not ices (I couldn't get it to compile; I'm > >using FreeBSD and can't code C worth a damn), so the second approach > >will not work. But I can write a parsing script (I had'n thought of > >the cue file), so I'll be trying that. > > Please don't use shout. It's buggy,

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Also sprach Robin P. Blanchard: > looks like you need to set > > CPPFLAGS="-I/usr/local/include" > and > LDFLAGS="-L/usr/local/lib" > > in your configure environment. Aha! That did the trick. Thanks, I'll be testing it presently. But why can't configure find the lame, shout and python libraries (all under /usr/local/lib), even with explicit

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have both ends set up with trunk=yes, notransfer=yes, type=friend. I notice that the trunking works from HEAD to 1.0.9 only (the direction in which calls are originated). I know this by bandwidth usage and by iax2 trunk debug. I did have to use trunktimestamps=no on the HEAD end to keep it quiet. I assume this is the new

-- Attempting native bridge of IAX2/peer1-iax-7 and IAX2/peer2-21 I want everything to stay in the VoIP server rather then briding. I have notransfer=yes on, but it still seems to bridge the call natively.. can I keep the RTP stream on the asterisk server some how?

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.11.1 DAHDI-Tools-v2.11.1 dahdi-linux-complete-2.11.1+2.11.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Notable changes: Raised E1

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.11.1 DAHDI-Tools-v2.11.1 dahdi-linux-complete-2.11.1+2.11.1 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete Notable changes: Raised E1

Hi, folks. I've installed icecast 1.3.11 and I'm very happy with it. But I'd like to add a "now playing" gizmo to my home page; can anyone point me to a php script or something else that does the trick? Thanks in advance. -- People don't quit playing because they ___vvz /( grow old. They grow old because they <__,` Z / ( quit playing. [Oliver Wendell

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the