search for: allowguest

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> exten=>6001,1,Dial...

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 a...

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](provid...

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=pe...

...om> 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 > alwaysauthreject=yes > ... I don't think either of these configuration attributes achieve what the OP requested ("how to Disallow CALLS without registering on PBX"). In fact, I'm not sure that it's actually possible to disallow [authenticated] calls from a peer th...

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 trunk. The Avaya engineer is telling me that I need to change my “From” header, and I don’t know how to do that. I have to figure this out in the next few days, or I’m in deep doo-doo. -T Thomas M. Peters |...

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 =>

...###### jabber.conf [general] autoregister=yes [asterisk] type=client serverhost=talk.google.com username=ldardini at gmail.com secret=********** priority=1 port=5222 usetls=yes usesasl=yes buddy=ldardini at gmail.com status=available ####### gtalk.conf [general] context=default bindaddr=0.0.0.0 allowguest=yes [guest] disallow=all allow=ulaw context=google-in [ldardini] username=ldardini at gmail.com disallow=all allow=ulaw context=google-in connection=asterisk ######## extension.ael context google-in { s => { NoOp( Call from Gtalk ); Dial(SIP/************@************,60,r);...

...[sipivr] host=dynamic secret=1NA6oZjTg1rjhZN8lArDgzLI7z8V2fxV type=peer ;context=transit-ivr context=incoming dtmfmode=inband The new section, with many failed experiments commented out, is after the [sipivr] section: [sessionmgr1] type=peer ;type=friend port=5060 host=10.90.0.17 dtmfmode=inband allowguest=yes qualify=yes realm=mcts.org promiscredir=yes ;Some have suggested using canreinvite=no with Avaya- didn't try that yet ;canreinvite=no canreinvite=yes transport=tcp ;context=incoming context=from-internal ;username=10.90.0.103 fromdomain=mcts.org disallow=all allow=ulaw allow=alaw tcpenable=...

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 call)/ <sip:+1...

...y situation: I'm running two Asterisk servers between two sites. They use DUNDi to route calls between the two servers so there is no dialplan to route calls between the two. Firewalls have been configured to allow SIP clients to connect from the Internet. I'd like to change my sip.conf to allowguest=no without having to recreate dialplans between the two. How is this accomplished? I can't seem to find much on using allowguest=no in sip.conf. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/a...

...39;ve enabled the DNS manager. Let's see how it goes. Kind regards, Sebastian > > > Am 14.04.2015 um 08:26 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > > >> On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote: > >> 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. > > > > Hello Daniel, > > > > Judging by the lists I found I think it's more like this su...

...affic (especially with qualify and multiple registrations) and they may even lock you out as untrusted, which may even result in that they will block asterisk permanently for everybody. Thats not really desirable. I think its also not recommended in terms of security to use default context with allowguest=yes and sort the incoming calls by header, because this can be faked easily. From my understanding the permit/deny parameters are only used for incoming calls if host is set to dynamic and then there will be no outgoing registration to remote peer possible. permit/deny is used for access, not...

...-------+ +------------------------------------------------------------------------+ | Description | Unauthenticated calls can be made via the SIP channel | | | driver using an invalid From header. This acts similarly | | | to the SIP configuration option 'allowguest=yes', in | | | that calls with a specially crafted From header would be | | | sent to the PBX in the context specified in the general | | | section of sip.conf. | +------------------------------------------...

...-------+ +------------------------------------------------------------------------+ | Description | Unauthenticated calls can be made via the SIP channel | | | driver using an invalid From header. This acts similarly | | | to the SIP configuration option 'allowguest=yes', in | | | that calls with a specially crafted From header would be | | | sent to the PBX in the context specified in the general | | | section of sip.conf. | +------------------------------------------...

...to PSTN fax doesn't work. Using udptl debug, I can see packets between Asterisk and both sides (SPA112 and PSTN fax) but it seems that faxes can't agree how to send image. == sip.conf: [general] tcpenable=yes videosupport=yes transport=udp,tcp dtmfmode=rfc2833 qualify=yes directmedia=no allowguest=no alwaysauthreject=yes rtcachefriends=yes rtupdate=no callcounter=yes t38pt_udptl=yes,redundancy,maxdatagram=200 t38pt_rtp=no t38pt_tcp=no ignoresdpversion=yes disallow=all allow=alaw allow=ulaw externip=82.200.7.184 localnet=192.168.0.0/255.255.0.0 [mtt] type=peer host=80.75.130.136 fromuser=749...

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 secret=123abcd context=internal [7002] type=friend host=dynamic secret=456abc...