similar to: additional range parameter for sip peer

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

Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip

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

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.

On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote: > 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

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

Scott, I have changed the configuration as said it and will test it. I?m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that

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>

Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: The peer ip address could be another than the ip address of incoming invites After an re-register the REGISTER is send to the new SIP server, answered with OK. But the peer ip address is still the old one (sip show peers). If now is a INVITE, the request is answered

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

On 4/1/15 10:48 AM, Daniel Heckl wrote: > John, > > thank you four your answer. I think you have misunderstood the > problem. It?s about a ip address change of the sip trunk, not of my > asterisk server. You would probably benefit by enabling the DNS Manager to allow for dynamic IP changes: # cat dnsmgr.conf [general] enable=yes ; enable creation of managed DNS

?I'd be curious if setting insecure=invite,port makes any difference either (without alllowguest on). ? On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl <daniel.heckl at gmail.com> wrote: > Ok, I have tested dnsmgr. This is not a solution, the situation has not > changed. With dnsmgr I can not place outbound calls. I do not know why and > what dnsmgr really do. > > My

On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: > 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. Hello Daniel, I'll find myself in the same situation a few weeks from now :-) > > For example, Asterisk sends a

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

Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer? > Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.heckl at gmail.com>: > > Hello Sebastian, > > I had already seen this list of the hosts, but it is not

We have quite a few frontend mail gateways that deliver incoming email trough our dovecot director (lmtp proxy) to the backend lmtp servers, and are seeing this logged on the frontend mail gateways: Sep 5 10:51:56 mailgw1 postfix/lmtp[23443]: 0E2F41C01A: to=<me at example.com>, relay=loadbalancers.example.com[192.168.42.15]:24, delay=61, delays=0.02/0.3/31/30, dsn=4.4.0, status=deferred

I just configured postfix' virtual_transport to point at my dovecot director, but am seeing occational problems like this: Aug 31 11:50:06 smtpgw postfix/lmtp[5339]: 69E2F5410D: to=<email at example.net>, relay=loadbalancers.example.net[192.168.42.17]:24, delay=0.15, delays=0.14/0.01/0/0, dsn=5.5.4, status=bounced (host loadbalancers.example.net[192.168.42.17] said: 501 5.5.4 Invalid

Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the new register goes out, it's creating a new state in the firewall, resulting in a new port number, which is why you would have to

https://issues.asterisk.org/jira/browse/ASTERISK-20841 The patch was already posted by someone but then was deleted because of guide lines. Is it really that hard to fix? Since 1.8 there is this problem but nobody seems to care about. Asterisk isnt only used with itsp who dont care about fromdomain. Or are the developers saying, we dont care about people who are using Asterisk in smaller

Scott, thank you four your reply. I had already though about both options, but the problem is, that after an ip change AND a new registration the ip address of the peer is not updated automatically. INVITES are answered with 401. Only after a sip reload the peer works again. That can't be normal... Daniel > Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <sgriepentrog at