similar to: How many calls can use the same username

Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi List; All we know that in voice, there are a type of communications between endpoints, for example: in some communications we do a proxy for media and signaling while other communications we do a proxy for only signaling. Where I can determine these things in Asterisk if I am using SIP and if I am using H.323? Regards -------------- IP Telephony and Contact Center Engineer Eng. Bilal Ghayad

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

Hi, Any one know why the glibc-profile is obsoleted on centos 5 (glibc 2.5)? I've checked the glibc.spec file and find that the --disable-profile option is enabled in the configuration command. So how the original profiling functions are implemented on glibc 2.5? Is there a new way to do the same/similar thing? Thanks a lot.

>Wine Is Not (an) Emulator. yup i know that :) >This means that it cannot run windows - it /is/ >windows as far as your >end-user applications are concerned. no issues with that >This means that you should not use your windows boot >as a 'fake >windows', but instead install applications properly >and use them. i usually run a setup of all my windows app in wine

--- Erik de Castro Lopo <erikd-flac@mega-nerd.com> wrote: > Hi all, > > I've got two problems compiling the current CVS FLAC sources on OSX. > > Firstly, the configure script can't find the OGG libraries which were > installed from MacPorts. I have tried: > > ./configure --with-ogg-includes=/opt/local/include \ >

Hello This is a bit off-topic, but here goes: Try installing the "device-mapper-multipath" package and follow the documentation below - residing at the powerlink.emc.com site: "Native Multipath Failover Based on DM-MPIO for v2.6.x Linux Kernel and EMC(r) Storage Arrays" When configured this works just as good as PowerPath. This works great on RHEL / CentOS 5 Daniel On

> From: Chuck Kollars <ckollars9 <at> yahoo.com> > Subject: Samba & ACLs? > Date: 2006-08-19 02:46:45 GMT > How exactly do Samba 3.x and ACLs interrelate? ... > I started out naively assuming that the *nix > uidNumber/gidNumber Samba mapped the end user to > would behave exactly the same whether they were a > Samba user or were logged on locally. ...

Thank you for you quick reply, Jean-Marc. I have just used a human voice to replace the sine wave, but the result is like a broken voice after tens of seconds. Do you have some sample voice in .sw that I can test with? I want to have one set or a few sets of sample voice that work for my understanding and debugging. This is the output from echo_diagnostic.m in my human voice test: Far end to

Hello list, I have defined a new MoH-class in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; *[106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes* In sip.conf I have this commented out : ;mohinterpret=default ;mohsuggest=default Asterisk sees these moh-classes and files : vps2301*CLI> moh show classes Class: default Mode: files

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

hello, the last 3 times I installed C5, the MBR was unchanged and nothing was written into /boot/grub except splash.xbm.gz there were no stage* files, nor a menu.lst I know how to fix that. Would I have better luck using a partition mounted as /boot? Anyone succeded with Grub that way? I prefer not having to do surgery to get C5 to boot :) -- Mark New Packages for C5 ---------

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

Hi to all i think this is not a new problem for this forum....but its newest for me as i m a new linux lerner. Even if it is new plzz....reply me ur answer..n if its already asked n have solution..plzz forward the solution. My problem is here mentioned: I have fedora core 4 as a linux server. there r two external links connected to this. the settings are as: eth0 ->for internal (that is

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Josh Coalson wrote: > > The test file is here: > > > > http://www.mega-nerd.com/tmp/flac_char.ogg > > yep, this is definitely a bug, thanks for the test case. requires > a tiny fix to the ogg seek algorithm which I will check in tonight. Oh cool. I was wondering what happened to that issue. Erik --

hi FLAC list! Does anybody know where I can request a new feature (if that's possible of course) for the FLAC frontend, included in the windows installer? The feature I would like (if the developers agree and want to implement it) is the possibility to add FLAC files to the list in the frontend and re-encode them to FLAC. This is useful if you want to re-encode some files by using the

Hello - I running Samba as a PDC on FC6 with roaming profiles. I need to setup a custom Windows logon/welcome message... to tell users want they can expect using this domain. Is it also possible to place different PDF files on the users desktop when he or she logs on, but only referencing one source file, so I don't have a copy for each user? What is the best approach? Thank You. Ralf