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    QUESTION HELP SETTING UP Virtual Hosted Server From Vitelity

    No sound is most likely a NAT issue. Ensure you have NAT=yes set on the extensions (try the different NAT options available). Also try Googling your local firewall/router and "SIP ALG". If your router does have a SIP ALG you could try turning it off.
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    RECOMMENDATIONS Monitor and auto restart Asterisk?

    I agree, find out why it's crashing would be best. You should be able to use monit to autorestart while you're doing that - https://mmonit.com/monit/ http://blog.tmcnet.com/blog/tom-keating/asterisk/using-monit-tool-to-monitor-asterisk.asp
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    QUESTION GSM phone as SIP extension

    I'm guessing you're not in the UK as you didn't say. If you were this works great - https://sysadminman.net/blog/2015/gsm-mobile-phone-freepbx-extension-sip2sim-6778
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    TIPS Yealink Security Concern

    I would check if there is a firmware update for your DSL/cable router. If the router is opening inbound port 5060 (SIP) from ANY IP address because there is an outbound connection from port 5060 to your Asterisk server then that's wrong. Certainly the BT router fixed the issue in later...
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    GO HERE (DTS) PROBLEM with IAX2-to-IAX2 trunk between local PiaF(s)

    I think that's a DNS issue. It's trying to resolve "raspberrypi2" to an IP. Should it be in your hosts file?
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    TIPS Yealink Security Concern

    If you don't see the calls in your Asterisk logs I'd check out this - https://sysadminman.net/blog/2014/insecure-home-routers-when-using-sip-6051 I've seen quite a few routers now that do this, not just the BT one in the post.
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    TIPS Where are my SIP packets going?

    Seems like your troubleshooting is good. You're outside the remote firewall, so it's not a SIP ALG messing up the packets. And the fact that other packets to port 5060 get through confirms it's not a general network issue. The only think I can think of is that your ISP is doing deep packet...
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    TUTORIAL Real time active call list?

    You're correct, I should have added some license info. I fixed this (under GPL/2) and took out the PHPAGI stuff. Parsing the output for the the fields is messy using the results of Asterisk CLI. You might need to check the 'core show channels concise' to get the correct fields. I think...
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    TUTORIAL Real time active call list?

    I wrote a simple web page to do this for myself - http://sysadminman.net/blog/2013/asterisk-outbound-call-status-page-5600 Not sure how "elegant" it is :-) It's just doing a "core show channels concise" and formatting the output, refreshing the table every few seconds.
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    TIPS Asterisk 100% CPU usage

    I've seen Asterisk do this when the astdb database is corrupt.
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    Choppy Sound and Poor Voice Quality

    The router could have been under load but as this line shows - 16. 69.147.236.82.rdns.ubiquityservers.com 0.0% 4.8 3.7 2.3 11.6 1.3 at the time the trace was done it wasn't affecting packets to the ultimate destination...
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    Choppy Sound and Poor Voice Quality

    Ping packet loss along the path does not necessarily mean there is packet loss to the destination. A router along the way might drop ICMP packets directed at it, but happily pass on ICMP packets to a different destination. That's not showing any packet loss to the ultimate destination -...
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    Linux Hard Drive activity.

    A little late maybe but for checking out what's causing disk access dstat is great. Available from the rpmforge repo for CentOS (5.5+ I thnk) - http://wiki.centos.org/AdditionalResources/Repositories/RPMForge Produces top process for disk access, amongst a ton of other things - dstat...
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    Skype?

    It's a shame Microsoft pulled the plug on this - http://www.digium.com/en/products/software/skypeforasterisk.php - when they purchased Skype, as it was simple to setup. I use it myself and so do some of my customers and it works well. Hopefully something similar will reappear in the future...
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    voicemail don't work

    You should check in the file /etc/asterisk/voicemail.conf, I've seen it get messed up a few times. Sometimes there are extra [ ] entries in there.
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    FreePBX 2.10

    The FreePBX team are continuing to improve FreePBX 2.10. The latest beta includes the automatic drop-down menus that lots of people asked for - freepbx.org/news/2011-10-10/2-10-update-and-come-say-hi-at-astricon-this-year I updated the demo system to reflect this -...
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    FreePBX 2.10

    Recently I’ve been searching for a web GUI for Asterisk that is much simpler than FreePBX to be able to offer customers as an alternative, for those that do not need all of the functionality. I really like FreePBX but it can be overwhelming when starting out as there is just so much of it...
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    FreePBX 2.10

    As far as I'm aware they're currently designed to be clicked on. They don't auto drop down in any of the browsers I've tried.
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    FreePBX 2.10

    Cheers Tony, I opened a ticket - freepbx.org/trac/ticket/5387#preview
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    FreePBX 2.10

    Besides the disabled modules things look pretty much the same on the FreePBX Distro. Speaking of disabled modules, it looks like there's a bug when creating a user account in FreePBX and then only assigning certain permissions. It generates an internal 500 error. Thought it might just have...
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