similar to: dnsmgr_lookup

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

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

?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

Hello, What's dnsmgr ? Anybody could tell mr more? cat /etc/asterisk/dnsmgr.conf [general] ;enable=yes ; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds ; default is 300 (5 minutes)serveur1:~# Harry

I "thought" I has everything using IP addresses. I am not making "outside" calls this is all internal. I have a connection between two machines both running asterisk. I am using 1.8.3 and I see a lot of dnsmgr_lookup's for "mymachine". I have a register line in sip.conf that is the only place mymachine is referenced. the actual definition for host= is the IP

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

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

Hello, We're attempting to use Asterisk for distributing calls via SIP in a large-scale speech recognition/VXML environment. We currently use DNS SRV with weights and priorities to instruct VoIP gateways (not Asterisk) to route calls to pools of servers. This works extremely well and provides for load balancing, fail-over, and by setting the TTL low (several minutes) we can easily take

According to my IAX-provider, an account has been created for me on their Asterisk-server... But the Asterisk CLI tells me this : asterisk*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found [Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring bindport on reload [Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10183 set_config: Ignoring bindaddr on reload

Hello list, I'm having DTMF-troubles with a Snom phone. I want to know if it's the Snom or Asterisk that makes the trouble. I'm playing a prompt, then make a choice for "2" : [Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] -- <SIP/test1-00000701> Playing '/var/lib/asterisk/sounds/vprompts/109001/prompt5040.slin' (language 'nl') [Jan

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

Hi Guys, Asterisk 1.6.2.7 install from Yum Repository shows a lot of : > doing dnsmgr_lookup for sip.provider.com Google searches show it was fixed in some version. Is this to be ignored? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100624/8e846c18/attachment.htm

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

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

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

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

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

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

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

I'm testing an upgrade of an i686 production machine running 1.4.24.1 to 1.6.0.9. I've installed dahdi-linux-2.1.0.4. But: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.