similar to: Refreshing DNS lookups

Displaying 20 results from an estimated 1000 matches similar to: "Refreshing DNS lookups"

2005 Jun 23
2
Asterisk 'losing' upstream provider registration state during small network outages.
Now that I have most everything actually working I've noticed that about every 3-4 days on average..... and at worse... Once a day my asterisk box seems to lose it's registered state with our sip provider and no longer will take any incoming calls. The caller simply hears a fast busy (reorder) If I do a reload at the command prompt all is well for another few days..... What I'm
2006 Feb 13
1
TDM04B/TDM2401E Card
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2006 May 15
1
Please..... need some help
Sorry if I post in this forum, this may be not the right one, but I hope to find in here some experts which could help me out. I have in one location 8 extensions from a Panasonic PBX ?KX-TD1232? connected to FXO Ports on an MultiVoIP Gateway from Multitech. On the other location I have 8 SP5100 IP-Phones from Micronet as remote extensions. Between the locations I have a VPN.
2009 Oct 30
1
asterisk 1.6 - doing dnsmgr lookup for... / call fails
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
2015 Apr 02
2
Update peer IP address
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
2009 Aug 15
2
bare minimum /etc/asterisk for sip based *
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
2007 Jul 02
1
Question about dnsmgr
[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
2015 Apr 02
3
Update peer IP address
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
2009 Dec 27
2
rxgain / txgain for iaxmodem or hylafax
In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never
2004 May 05
2
connect a sub telefon system?
<P>Hi List,<BR>is it possible to connect an existing telefon system with a stanard AVM ISDN card to the telco?</P> <P>&nbsp;</P> <P>[ISDN from Telecom supplier] ---- [Aterisk Box] ----- [ existing telephon installation]</P> <P>&nbsp;</P> <P>kind regards,<BR>Patrick</P>
2015 Feb 05
2
IAX2 problem for WAN connections
Hi, I am trying to connect two Asterisk servers using IAX2. Everything works fine when I couple them within a LAN segment, but not when I connect them using WAN connections. I made sure that the routers' ports are mapped properly and checked this with additional ssh rules. ServerA is a Raspberry box with the vendor's Asterisk version 1.8.13.1 and ServerB is normal CentOS 7 box with
2015 Apr 01
2
Update peer IP address
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
2020 Jun 07
3
CDR mysql: timeout when remote database unavailable
> On 2020-06-06 10:38, Antony Stone wrote: > On Saturday 06 June 2020 at 09:18:11, Fourhundred Thecat wrote: > >> In a situation when I start asterisk, and the remote database is >> unreachable, asterisk waits for several minutes before it actually >> starts (before it loads sip module, etc). >> >> And when database is unreachable during operation, when call
2006 Jan 10
2
Problem with Action:Originate with ASterisk Manager
Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:
2007 Sep 18
2
asterisk crash and core dump
My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51
2005 Aug 29
2
Compile problem with 1.2 beta 1
Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: 1) No-one else is trying beta 1 2) No-one else is having any issues (I must be the idiot) 3) No-one else saw my message :) I have been trying to compile 1.2 beta 1 on a centos 4 box, to no avail. The "make" command seems to compile ok, but "make install"
2003 Mar 09
6
DTMF detection on SIP provider ?
Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike
2012 Feb 11
1
New router, registration problems
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
2010 Sep 13
3
doing 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
2007 Apr 30
2
Improving Asterisk's DNS support
Hello everyone, After several years of using Asterisk I have always been frustrated by the support for DNS. I have seen all kinds of strange behavior when Asterisk is used on a system with "iffy" DNS servers: - no failover to other DNS servers in /etc/resolv.conf (might be a C library thing) - chan_sip will sometimes mark even local SIP peers as unreachable during/after any DNS