similar to: Asterisk Maint.

Hello. I have been searching the archives for a simple, clear listing of the available codecs with total size, plus the data and overhead sizes. Does anyone have this handy, and can it be added somewhere, even the wiki. Regards...Martin -- The system will be down for 10 days for preventive maintenance.

I would like to know how to calculate the amount of bandwith I would need to host X number of calls. For example, if user A in San Francisco with an ATA 186 calls user B in New York with an ATA 186 and Asterisk is being hosted in a PC in Miami. How much bandwith do I need to have in Miami? Do I just need bandwith for the setup of the call (ie the SIP part) or are there any instances where the

Hello, I am very new to R (read the introduction a few hours ago), but experianced in several other languages (including Scheme, C, ObjC and C++, Java and Javascript) and I was wondering how to approach this problem: I am making a training calculator for a Video Game I play. Basically, I want to calculate the probability of killing an enemy within x hits, to help determine which is the best

Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

As discussed at: https://www.redhat.com/archives/libguestfs/2014-September/msg00215.html Signed-off-by: Chen Hanxiao <chenhanxiao@cn.fujitsu.com> --- maint.mk | 7 ------- 1 file changed, 7 deletions(-) diff --git a/maint.mk b/maint.mk index b6ec1b5..76759d4 100644 --- a/maint.mk +++ b/maint.mk @@ -408,13 +408,6 @@ sc_error_message_uppercase: { echo '$(ME): found capitalized error

> -----Original Message----- > From: Richard W.M. Jones [mailto:rjones@redhat.com] > Sent: Thursday, March 12, 2015 4:32 PM > To: Chen, Hanxiao/陈 晗霄 > Cc: libguestfs@redhat.com > Subject: Re: [Libguestfs] [PATCH] maint.mk: remove error_message_period check > > On Wed, Mar 11, 2015 at 01:18:45AM -0400, Chen Hanxiao wrote: > > As discussed at: > >

On Wed, Mar 11, 2015 at 01:18:45AM -0400, Chen Hanxiao wrote: > As discussed at: > https://www.redhat.com/archives/libguestfs/2014-September/msg00215.html > > Signed-off-by: Chen Hanxiao <chenhanxiao@cn.fujitsu.com> > --- > maint.mk | 7 ------- > 1 file changed, 7 deletions(-) > > diff --git a/maint.mk b/maint.mk > index b6ec1b5..76759d4 100644 > ---

Hi Rich, I noticed that libguestfs's cfg.mk had an old (and relatively inefficient) version of the rule that checks for empty lines at EOF. The one in maint.mk is much more efficient: >From cf93f04a980ec01ef37db9a0d4c7b99015169eee Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Mon, 17 Sep 2012 11:49:34 +0200 Subject: [PATCH libguestfs] maint: remove a

>From 34af5f3be8b04443dc151e6d070aa49cbe59b7fc Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Fri, 20 Nov 2009 11:50:54 +0100 Subject: [PATCH libguestfs] maint: remove unnecessary include of openat.h * daemon/realpath.c: Don't include "openat.h". not used. --- daemon/realpath.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Does hivex have a trivial-patch policy? >From e23c45a2f813ac29b022faa9112c1977b651862f Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Tue, 28 Aug 2012 17:34:26 +0200 Subject: [PATCH hivex] maint: remove unnecessary mkdir -p * ocaml/Makefile.am (install-data-hook): Remove unneeded mkdir -p. It's job is handled by the very next mkdir -p. ---

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

The Wiki page has been SSL-only for a few days to a week or so. [perhaps this is by design, I don't know - but it is different than it was a week or more ago.] But the link to it from the main samba.org page is wrong and the suggested link doesn't get you to the wiki either. [It goes to CIFS.ORG.] Most of us can find our way - but it probably needs addressing sometime soon. -Greg

Tell automake that we want ./nbdkit to be kept up-to-date in the developer's tree. Otherwise, after touching nbdkit.in, you would have to remember to manually run 'make nbdkit'. Signed-off-by: Eric Blake <eblake@redhat.com> --- Makefile.am | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Makefile.am b/Makefile.am index 315aa36..7815258 100644 --- a/Makefile.am +++