Re: [asterisk-users] Recurring SIP problem with asterisk 11.6 11.7
On Wednesday 13 November 2013, Jeremy Kister wrote: On 11/12/2013 8:46 PM, Duncan Turnbull wrote: Any chance DNS is dying about the same time the problem occurs good idea, but I don't use DNS anywhere in Asterisk. well, except for sip.conf:externhost. it's all IP addresses. That doesn't matter. Without a working DNS server, Asterisk will fall over -- even if all your configuration files are using IP addresses and you have all your own machines in /etc/hosts . Probably the easiest thing to do is just install bind9 on your Asterisk box (and make sure to put nameserver 127.0.0.1 in its resolv.conf). BIND is pretty lightweight as servers go, so should not impact badly on performance, especially if it's the only machine using it. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recurring SIP problem with asterisk 11.6 11.7
Throwing in my my 2cents, I prefer dnsmasq, which is even lighter and Asterisk doesn't mind. As far as maintenance (firmware updates of phones, etc) goes, dnsmasq also offers TFTP and DHCP functionality. If nothing like that is running on a customer site, this is quite handy. Not that you can't do this with other packages, but dnsmasq is very easy to configure. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
I frequently use Audacity to analyze the audio data. In many cases I can see from the spectra (and other graphical representations) with what kind of problem I am dealing. Meanwhile, for most of my problems I no longer depend on an audio editor. I don't know whether this is helpful in your case. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP sound quality : highroad sound
2013-11-13 11:55, Jonas Kellens skrev: On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Here is some suggestions to get started: http://www.enterprisenetworkingplanet.com/unified_communications/troubleshooting-common-sip-problems-with-wireshark.html Maybe one of your connections get congested? For example, if the two endpoints is your phone and the upstreams teleco. If the side from the teleco are bad and not the phone you need to take a closer look at the switches and routers on the way to the teleco. For example you can run tcpdump on your gateway to your ISP. If you see the problem here as well it may be your link or a upstreams problem. If you don't see it here it is somewhere in between.. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calendar.conf include
Hello, can I use include-statements in the calendar.conf configuration file ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Mass exodus
Hi all, I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints aren't on the same networks. We don't see excessive CPU load when it happens. It does SEEM to happen most right after someone accesses their voicemail. We are using RT SIP registration as well as database voicemail storage (mysql). The database is on the same machine as the asterisk server. Have we grown beyond the ability to host both the db and * on the same hardware? Or is this a known issue with a (hopefully) known fix? TIA, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Presence across two servers
Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and fault tolerance but we have a unified dialplan and SIP trunking from site to site via our VPN. Everything presence related works wonderfully for local users, but I'm hoping there's a way we could get presence for the people at the other end of the pipe fairly transparently. We have a lot of cross-office collaboration, and our office manager/receptionist (who has the battleship of a 7961G+7914+7914 BLF) would love to at a glance know if the remote folks are available for a call or not. I'm sure this has been covered, but my Googlefu us turning up a ton of redundant, old, and deprecated information so I've resorted to asking here. From what I have found it sounds like it may be easier with IAX2 but my experiments with IAX2 haven't yielded wonderful results and management prefers SIP everywhere If anyone has any pointers I'd greatly appreciate it - thanks in advance! Lincoln *- One of the worst IT decisions I've made for better or worse. Looked good on paper; in practice not a good idea for anything beyond a very simple SOHO. -- Lincoln King-Cliby, CTS, DMC-D, CCMP-S Commercial Market Director Sr. Systems Architect | Crestron Certified Master Programmer (Silver) V: 440.449.1100 x1107 F: 440-449-1106 I: http://www.controlworks.comhttp://www.controlworks.com/ Crestron Services Provider -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Mass exodus
On 13 Nov 2013, at 18:29, Mike Diehl mdiehlena...@gmail.com wrote: I've been seeing some strangeness lately on my 10.2.1 server. It's gotten to the point that a few times each day, I see masses of SIP clients becoming unreachable. They're not all on the same network, and we don't see any calls drop. In a few seconds, they all come back. I don't think it's a connectivity issue because we don't drop calls, and the endpoints aren't on the same networks. We don't see excessive CPU load when it happens. It does SEEM to happen most right after someone accesses their voicemail. We saw this happen on a 1.4 server a couple of years ago shortly after 2am each day. It was only after a study of the cron schedule we narrowed it down to a number of rsync backup jobs which were run at that time. As in your case, it wasn't a connectivity or bandwidth issue - in the end we put it down to a disk I/O bottleneck. It might be worth running something like iostat on your box to see if you see a spike in iowait as voicemail is being checked. We resolved it simply by rate limiting our rsync jobs. In your case with a busy database, you might want to look at your MySQL indexes and/or cache settings - this might be something worth asking about on the respective MySQL discussion groups as well as here. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI version vs. AST version
Is there a mapping of AMI versions to Asterisk versions somewhere? For example, Asterisk 1.4 includes AMI version 1.0 (at least that's what I see when I connect to Ast 1.4 via telnet to the AMI port) Also, doe the AMI version changes reflect changes to the AMI commands? If so, is there also a list of what commands changed by AMI version? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Mass exodus
Mike, Am 14.11.2013 01:48, schrieb Chris Bagnall: As in your case, it wasn't a connectivity or bandwidth issue - in the end we put it down to a disk I/O bottleneck. It might be worth running something like iostat on your box to see if you see a spike in iowait as voicemail is being checked. We resolved it simply by rate limiting our rsync jobs. In your case with a busy database, you might want to look at your MySQL indexes and/or cache settings - this might be something worth asking about on the respective MySQL discussion groups as well as here. you could also throw in some more stuff like the following and run it for 24 hours every second, then check the log for the timestamp to figure out what happened: while true; do echo `date`; asterisk -rx 'sip show peers'; asterisk -rx 'sip show channelstats'; mysqladmin --password=yourpass processlist; ps auxww; vmstat; iostat; echo; echo; echo; sleep 1; done log.txt Regards Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] e1 , hdlc data link?
Hello! I want to use TE121 for E1 data link to cisco. Really only for tests now. So I wrote: span = 1,1,0,ccs,hdb3,crc4 #span=1,1,0,esf,b8zs nethdlc=1-31:hdlc0 may be first line is wrong, problem is somewhere else anyway. Just because I get: # dahdi_cfg DAHDI_CHANCONFIG failed on channel 1: Function not implemented (38) This is centos 6.4 box with dahdi 2.7.0.1. Could you tell me is it possible? Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
