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This has come up before on the list and archives but I don't seem to find a solution for this. On just a few nodes we have this situation where we see the IP disappear from the CLI iax2 show peers list but the status shows OK: 3012/3012 (Unspecified) (D) 255.255.255.255 0 OK (89 ms) How can the status be OK a few milliseconds ago and have no IP ?? The strange thing is

...the scheme would be something like: PSTN <------> FXO ----- | |------------Asterisk | FAX ------> FXS --------- I'm using Asterisk 1.4.26.2 on FreeBSD 8.0 TIA, Alejandro Imass

...I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xxxxxx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=101 at home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten => 101,1,Dial(SIP/101,15) exten => 101,2,Voicemail(101 at home) ...sn...

Server will have two fix public ips. Dave > -------- Original Message -------- > Subject: Re: [asterisk-users] load balance with 2 wan connections > From: Alejandro Imass <ait at p2ee.org> > Date: Sat, December 25, 2010 1:58 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > > On Sat, Dec 25, 2010 at 1:18 PM, dave george <dgeorge at teletoneinc.com> wrote: > > Need some ad...

Hello, I have a Sip phone (Siemens C470IP) which works perfectly with different VoIP providers (iptel, betamax, ovh...). It also worked well with my testing server (ubuntu and inside the LAN). But now the problem i have is that the hardphone doesn't connect to my dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing is that ekiga can connect to the same asterisk server with

Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not

...FBSD (75/5 respectively). I personally think that there must be at least 50 million Linux/FBSD machines in the world in use today, easily (maybe even double that). If 1% of the servers use Asterisk that would be 500K. With the numbers above, I think it must be between 300 and 500K. Best, Alejandro Imass

Hi, I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 things: 1) allow any Asterisk SIP extension to call any Skype "user". I do not need to call landlines via Skype. 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and route the call to a specific Asterisk SIP extension. At first, I thought it would be

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has