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...The first one is probably a firewall, but the 147.135.0.129 indicates a larger problem. P.S - Can phones that do not support outbound proxy also register at 147.135.8.128 (not just Asterisk?) ----- Original Message ----- From: "Wolfgang S. Rupprecht" <list+asterisk-users@lists.wsrcc.com> To: <asterisk-users@lists.digium.com> Sent: Monday, July 26, 2004 1:57 PM Subject: Re: [Asterisk-Users] Broadvoice problems again > > oej@edvina.net (Olle E. Johansson) writes: > > The easiest first-level hack would be to randomly choose on of the > > SRV records p...

...a context that prepends "393-9" to their 6-digit prefixes? (And for extra credit "393-99" to their 5-digit prefixes?) Unless I can fix up their CID, the dialback buttons on the phone (and in voicemail) are useless. -wolfgang -- Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/ Send personal replies to this address. Mailman won't let me post unless I forge the From-line to be the same as my incoming alias for this list. <wolfgang+gnus20040629T111236@dailyplanet.dontspam.wsrcc.com>

Hi everybody, Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? ("/var/run/", "/var/log/asterisk/messages") any help will be appreciated, Cyprien

Hi, I compiled asterisk and chan_h323 on an Opteron in 64 bit mode. In the h323's Makefile I replaced in line 24 CFLAGS += -march=$(shell uname -m) by CFLAGS += -march=k8 and also tried CFLAGS += -m64 -march=k8 Both solutions do compile, but when starting asterisk, a load error occurs: undefined symbol: _ZN14H323Connection24OnUserInputInlineRFC2833ER15OpalRFC2833Infoi When I grep

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

I had my asterisk configuration working very well with broadvoice, but it stopped working this afternoon. I plugged the Cisco 7960 phone I used for my origional signup (just a few days before they offered generic BYOD) and it works fine. I did notice it seems to do all of its comunication through proxy.broadvoice.com (I used tcpdump). I have never contacted broadvoice about using asterisk

http://yro.slashdot.org/article.pl?sid=04/08/04/2212251&tid=158&tid=95&tid=103 Probably some of you already saw this. Now, beyond discussions regarding the legitimacy of such a ruling (whether they have the legal, moral or whatever right to enforce it), there's the technical aspect. Suppose i provide VoIP services using Asterisk, and i fall under the incidence of the FCC ruling

I was hoping someone can help with a problem with my GrandStream Budgetone "hanging" after a while. Problem seems related to the SIP registration - I am using dynamic registration (host=dynamic in sip.conf). Static IP is not an option in my case. I start Asterisk and all is groovy, phone works fine and can dial around. Then, at the time specified on the phone in the "Register

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

...P, & I don't have $2000 > for a T1 test set. > > Thanks, > Glen > > > > --__--__-- > > Message: 7 > To: asterisk-users@lists.digium.com > Date: 12 Jul 2004 09:14:36 -0700 > From: "Wolfgang S. Rupprecht" <list+asterisk- users@lists.wsrcc.com> > Organization: W S Rupprecht Computer Consulting, Fremont CA > Subject: Re: [Asterisk-Users] permission problem > Reply-To: asterisk-users@lists.digium.com > > > cyprien.simons@tu-berlin.de (Cyprien Simons) writes: > > Is the only way to use asterisk _not_ as roo...

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

This patch against OpenBSD -current adds a simple form of PKI to OpenSSH. We'll be using it at work. See README.certkey (the first chunk of the patch) for details. Everything below is BSD licensed, sponsored by Allamanda Networks AG. Daniel --- /dev/null Wed Nov 15 15:14:20 2006 +++ README.certkey Wed Nov 15 15:13:45 2006 @@ -0,0 +1,176 @@ +OpenSSH Certkey + +INTRODUCTION + +Certkey allows

It looks like asterisk isn't honoring EOF on stdin. file add-phonelist: database put cidname 200551234 "name 1" database put cidname 200551235 "name 2" database put cidname 200551236 "name 3" database put cidname 200551237 "name 4" asterisk -rn < add-phonelist What I see is an infinite stream of prompts as asterisk is banging onto the EOF. How