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This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom root@localhost.localdomain Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: root@localhost.localdomain (Cron Daemon)

I want to share some facts with the Asterisk community. I have been very successful providing a Vonage-type system based on Asterisk. For instance, one company that uses Asterisk and offers a similar service to Vonage is Voyze.com. The key concept is that Asterisk works like a Cisco, for all the intelligence is provided by SQL Server, outside Linux. I don't even save the CDR locally. The

...dres) > 26. Re: H323 (Tzafrir Cohen) > 27. Re: polycom soundpoint ip 300 (harry gaillac) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 27 Jun 2005 22:33:54 -0400 > From: "Federico Alves" <sales@minixel.com> > Subject: [Asterisk-Users] Asterisk ended with exit > status 1 > To: <asterisk-users@lists.digium.com> > Message-ID: > <200506280233.j5S2XvWo008812@ylpvm01.prodigy.net> > Content-Type: text/plain; charset="us-ascii" > > I need some brain-help:...

When the inbound leg of the all is SIP and the outbound leg is Oh323 (Voip-to-Voip only here), the DTMF relay (either RFC2833 or SIP Info), fails to go through, while it works perfectly when both legs of the call are SIP. Is this a shortcoming of the Asterisk core or the Oh323 channel? Is this solvable at all with some configuration change or a simple rewriting of the Oh323 channel driver? Second

I need some brain-help: I installed the chan_h323 software, and if I start manually Asterisk either by typing safe_asterisk or simply asterisk, it works, but it fails to start when I insert safe_asterisk or simply asterisk in /etc/rc.d/rc.local. The asterisk service script also fails. AsteriskAutomatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1.

Am I reading the facts wrong or SIP will receive calls from unauthenticated SIP user agents? The phones don't have to try to register. Of course autocreatepeers=no. Is this behaviour by design? If the phone tries to register than the attempt will be rejected, but the phone can still send calls to Asterisk, and they will be processed. If that is the case, why do I need to use SIP authentication

I can change the sip port to any number, and when I unload and reload chan_sip.so, I always get == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 64.1.16.172:5060 == Using TOS bits 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode' Is

I am blocked on the most simple step when compiling Oh323: Cd /openh323 patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch it hangs for ever and never finishes in my machine with Red Hat Enterprise Linux 3, fully updated. I need to use Asterisk HEAD and I therefore I want to use asterisk-oh323-0.7.2-pre1. Additionally, because it is the most debugged version. In the past, if

Before I go ahead and spend $40.000 on a Cisco 5400, just because my clients need T.38 faxing, I want to ask the community if there is any chance of having Asterisk receive G729+T.38 and sending the call via Zaptel to its final destination. Any answer will be appreciated. Federico

Four years ago, I faced a real dilemma in my business: the Visual Voice PRI dll had a bug that considered unanswered any call after ringing for 20 seconds. This bug was in fact killing my business, because for international calling, the setup of the call was already close to 20 seconds on many cases. Furthermore, the vendor, Artisoft, had cowardly sold the software to Dialogic, and Intel-Dialogic