similar to: realtime peer w/ callbackextension does not register after 'sip reload'

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

?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

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

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

That sounds like asterisk was working 100% correctly. If you receive an INVITE from an unknown IP address, then it should fail. Unless you want to allow anonymous, which is genearlly a very bad idea. If you are registering to IP X, but the provider may be transmitting invites from any number of other IP addresses, then you need a list of IP addresses, and have a trunk configuration set up for

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

Hi I have an installation based on Daniel-Constantin Mierla's excellent Kamailio 3.3 / Asterisk 10 Realtime document ( http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) but have come across an issue which is a potential problem. In this installation all SIP clients register with Kamailio, and the registrations are forwarded to Asterisk. This means that all

Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI> reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

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

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging again in the CLI> on host15 10.10.10.15

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

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

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

On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: > 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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 On 06/08/2015 01:18 PM, Luca Bertoncello wrote: > Hi list! > > Another day, another problem... I'm checking with Nagios my > Asterisk at home, and since yesterday I noticed that, after my IP > changes (Deutsche Telekom drop the DSL-line every 24 hours, so that > I have a new IP every day), the peer of an VoIP-provider I use is

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.

I just jumped to asterisk-1.6.1.8 and I calls will not go through to my asterisk. Same setup with asterisk-1.4 and calls get accepted. sip show registry (asterisk-1.6): Host dnsmgr Username Refresh State sip.actio.pl:5060 N 4589835 105 Registered sip show registry (asterisk-1.4): Host Username Refresh State sip.actio.pl:5060 4589835

Hi all. Trying to setup H.323 via Asterisk between a PLANET H.323 box and my SIP phones. When calling from the SIP phones, it connects but quickly disconnects citing the following error message: **** --- build_peer +++ build_peer +++ reload_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data