similar to: doing dnsmgr_lookup

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

I am using 1.8.3 and changed enable=no on dnsmgr.conf - however I am still getting log messages for dnsmgr_lookup. I wasnt expecting that. I have a server and a couple dedicated machines just running ALSA connections. I dont need any dns lookups for anything - who do I disable it? Thanks jerry ------------ Asterisk Ready. *CLI> *CLI> > doing dnsmgr_lookup for 'mndemo'

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

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

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

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

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

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 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 list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

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

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

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

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

Ever since upgrading to 1.6 I get messages like these. I want everything else that shows up, but is there a way to make all the dns messages go away? Ira > doing dnsmgr_lookup for 'gw5.telasip.com' > doing dnsmgr_lookup for 'sipconnect.ipcomms.net' > doing dnsmgr_lookup for 'proxy.ideasip.com' > ast_get_srv: SRV lookup for

If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying http://checkip.dyndns.org and compares it to the IP address stored in the parameter ?externip? in the [general] section of sip.conf. If the two values are the same, the script exits quietly. If they are different, the script updates

What files at a bare minimum need to be in /etc/asterisk for an asterisk server that does sip only and voicemail. I'm setting up an asterisk server to provide service for a single SIP softphone extension with SIP origination and termination. The main purpose of using * is for voicemail and future expansion ability. I know I need sip.conf extensions.conf voicemail.conf but what else? do I

[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to

I just set up a WRT54GS and now I can't dial out or in. sip show registry shows: CODE: SELECT ALL Host dnsmgr Username Refresh State Reg.Time draytel.org:5060 N xxxxx 120 Request Sent I seemed to recall that running in cli always showed a response back, but there's nothing now. Using

Hi all, I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the "host" parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a "iax2