similar to: Asterisk 1.6.2.13 - have asterisk reply from same IP address

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone....!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference,

Hello Sebastian, I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed. A port scan, to eventually update the list, found hundreds of servers provided in the address range 217.0.0.0/13 with open port 5060, some were even not found. I think there must be another solution. If I change insecure to

I'm looking for a way for linux to query a pc if user X is on, and has used the pc recently or the screensaver is not active. If so, I'll route a call for user X to the phone near that PC. Ideas, anyone? Leif

Hi List; How can I configure asterisk to receive a call from SIP end point without being registered at asterisk and its IP address is dynamic, and authentication to be based on the username and password or any other string? I know that if I place the host with static IP then no need to register, but what if the voip gateway was having dynamic IP and I do not need to register on asterisk, but I

You have two options for dealing with an IP change during the registration period: 1) set the registration time to shorter period of time to minimize the downtime 2) detect that the IP address has changed via whatever method available, and then issue a "sip reload" CLI command to asterisk, which will cause it to resend registrations immediately. On Tue, Mar 31, 2015 at 1:36 PM, Daniel

Dear All Can you please let us know how we can modify our Asterisk "extensions.conf" file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one

On Tue, 2004-05-11 at 21:52, Leif Madsen wrote: > Afternoon all, > > Jared Smith and I would like to have a conference call Sunday evening to > discuss the layout and direction of the Asterisk documentation project. We > both feel that the layout we have is a good start, but it needs to be > revised. I would consider what we have so far a first or second draft. > What we

Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable. For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de <http://tel.t-online.de/>), the message is answered with OK and the peer is registered. Usually INVITES comes now from this ip address. All

I want some cheap ip-phones with auto-answer, to work as paging system at dinnertime. Options, please. Leif

Is there a way to specify which IP address to originate calls from in a peer on sip.conf? I need to send calls from 10.1.3.10 which is a routed network through openvpn, but it's using 10.39.0.10 which is a vpn IP address - the asterisk box is the same box as the vpn bridge for the 10.1.3.0/24 network. I can't set the host as 10.39.x.x as it is dynamic. i can't change bindaddr since i

I'm trying set up asterisk on an amazon instance in Sydney. It's to use for our kids in Sydney to connect with their friends in the States. We've found iax works better than sip with these distances. But we now have weird problem: everybody has a cell phone, and it's much cheaper/better to use the house internet connection over the phones wifi. Each cell phone has it's own

Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able

Hi, my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when user and asterisk are on Local LAN. But my asterisk server is on public ip and users are behind nat, and so this method says unknow host when used on public asterisk server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel.

I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a

I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Fri, May 5, 2017 at 4:29 PM, Nicolas Bock <nicolasbock@gmail.com> wrote: > Hi, > > I am running a webserver on the libvirt host and would like to add a > nwfilter such that a VM can access that server. The corresponding iptables > rule would look like this: > > iptables --append INPUT --in-interface virbr0 --destination 192.168.122.1 > --protocol tcp --dport 80