search for: allowguests

Hello, I want that people from other servers like ekiga.net can make calls to my users. When I do an "allowguest=no" then people from other domains cannot call me. So I think I need "allowguest=yes". Maybe something like this? ------------- <default> include => users <dialout> include => users exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT) <users>

In your sip.conf file allowguest defaults to yes. This means that anyone that can reach the SIP ports on that system has access to make unauthenticated calls, by default. The administrator actually has to go in and turn it off to prevent unauthenticated SIP calls (in whatever context [general] points at). Does anyone else agree with me that this is a poor default? I'd like to see the

HI ALL got small question i use call-limit=1 on peers but call limit is not working if user is not registered on PBX and making calls so the main question is -- how to Disallow CALLS without registering on PBX -- Best regards Antony tel. +380669197533 tel2. +380636564340 Paypal http://paypal.me/Satskiy

I see in my log file this: Jun 30 21:44:26] NOTICE[42192][C-000002f3] chan_sip.c: Call from '' ( 5.189.144.120:5076) to extension '011972592675431' rejected because extension not found in context 'default'. which is great its rejected - however in my sip.conf file I have deny=0.0.0.0 permit=x.y.z.z/255.255.255.255 permit=a.b.c.d/255.255.255.255 So I'm expecting to

I have some problem finding a smart way to add inbound trunks ip authentication. I don't want to set allowguests=yes Some of my providers just list some IP and I add them like: [provider](!) context=fromoutside type=friend insecure=port,invite disallow=all allow=g729 allow=ulaw allow=alaw canreinvite=no [magrathea1](provider) host=87.238.72.129 [magrathea2](provider) host=87.238.72.130 [magrathea3](provide...

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no >

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip

All: I've starting building an asterisk system for our company, which has about 60 users. I am new to asterisk, so thank you for your patience. I've stripped the sip.conf and the extensions.conf down to the bare minimum: Here is my extensions.conf file [globals] [general] autofallthrough=no [default] [fromprovider] exten => YYYYYYYYYY,1,Dial(SIP/151,20) [phones] exten =>

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="

Hello all, I hope someone can help me with this old Asterisk version. I have to run this version because of a custom IVR written on it. Porting it would take much too long and we'd have to hire a consultant because of all the hooks it has into Oracle databases and real-time information. We have a brand-new Avaya phone system in place and we will be cutting over to it in late March 2019.

OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand

Gents, this concerns a CVS-HEAD downloaded today. I configured my system as I usually do, including using allowguest=yes to attempt to correct the following problem, but to no avail. When any call comes in from an external server I get this: Oct 1715:36:43 NOTICE[4040]: chan_sip.c:10774 handle_request_register: Failed to authenticate user "+16143691415" /(this is the number making the

Hi, I recently found someone was using one of my Asterisk servers to make international calls via some SIP method that allowed them to bypass authentication (running 1.4.21.1 so I'm not sure how they did this since the major vulnerability for this was patched in 1.4.18.1). At any rate I caught it the same day they started this, so I've blocked their IP range and put in some monitoring

On Tue, Apr 14, 2015 at 09:38:22AM +0200, Daniel Heckl wrote: > Sebastian, > > Your code sounds good, I'm curious how it goes on. > > First the linux machine had the Google Public DNS 8.8.8.8 as DNS > server. After I changed it to the via PPPoE assigned DNS servers, i > had no changes any more. But we should be prepared for changes. > > You must enable the dnsmgr.

Many ITSP are using loadbalancers, so if somebody registers on a sip peer with specific dns host, an incoming call may be received from a different ip and the host value in peer section doesnt match, so it will go to default context. For example Telekom or 1&1, biggest providers in Germany are using too many different addresses that its not practical to define them all (up to 50 hosts

Asterisk Project Security Advisory - AST-2008-003 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Unauthenticated calls allowed from SIP channel | | | driver

Asterisk Project Security Advisory - AST-2008-003 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Unauthenticated calls allowed from SIP channel | | | driver

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic