similar to: incoming cid translation tables

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

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

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S.

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

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

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

Hi, does anybody know how to decode/recode sambaMungedDial attribute? I need to store the terminal server profile path in it but can not find any documentation. Thanks, Alex

I have a samba ldap pdc set up. (2.2.8a). I have a windows domain member that is joined to the domain running ras. When users try to dial in to the server they get "Does not have Dial in permission/rights". Is this an option that samba recognizes at this point? Is there a way to tell windows that a user has dial in privileges? Sean Cook Kinex Networking Solutions

---------- In?cio da mensagem original ----------- De: asterisk-users-admin@lists.digium.com Para: asterisk-users@lists.digium.com Cc: Data: Mon, 12 Jul 2004 11:48:05 -0500 Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide

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 =

I patched samba to always return ACCESS_GRANTED for testing. So I came to this: IASSAM.LOG [556] 23:23:53:671: Inserting attribute msNPAllowDialin. [556] 23:23:53:671: Successfully retrieved per-user attributes. Dialin now "only" fails with "Dialin not allowed for user", but I'm not able to set it in UserMgr. Is it difficult to map this attribute? Daniel

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

I am unable to get any softphone to register to my asterisk server when I am connected via VPN. I have tried Ekiga, LinPhone, and Twinkle... on multiple machines. It works fine when locally connected (same subnet). The VPN is not NAT'ing anything... and all other connections work fine across it (i.e. http, ssh, scp, ftp, etc). In fact, the asterisk logs show the connections, so its getting

Hi all, We've been experimenting with the app_queue application, and it works quite well. The only problem we encountered was that outgoing calls (to the operators) aren't logged in CDR. Example, * operators dial a specific number/extension, and AddQueueMember(..) runs (they get added without any problems), and they Hangup. * normal users dial the support/hotline number, get added

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I know I'm missing something obvious, but I cannot wrap my wits around this one. I've been staring at it for too long I think. Maybe it's the three am syndrom! : ) So a call comes in and my snom ends up with this entry: CALLER NAME <sip:1231231234@server.ip> under missed calls, or whatever. Now I want to just click OK and