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  1. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun AWESOME!!! Mikrotik routerboards are fantastic good choice. They have great QoS options as well. For billing i would say try looking at Simmrates. It intergrates with Elastix and is easy to configure. Glad i could help. :lol:
  2. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Just google "step by step install openvpn server on centos" You can change your ports but i dont recommend that because you have to change it on all your devices as well. There is nothing wrong with the default ports. RTP is a UDP...
  3. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Hi I have never setup TLS or SRTP but apparently Asterisk needs to be recompiled with the SRTP support. Basically when you have the asterisk source and do the following: ./configure make menuselect. This is where you would enable SRTP...
  4. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun There are ways like AsteriskTLS and SRTP but it gets complicated and not all devices support it. The best and easiest is VPN im afraid.
  5. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Looking at your screenshot jitter buffer sometimes cause more problems that it solves so I would just disble that.
  6. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Ok. I would hide that IP and delete the posts on here that show it becuase I checked and its wide open for anyone to see. These are the settings you should look at in SIP settings. Try changing the setting from public ip then static...
  7. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun This comes back to me saying that there is maybe no route back to your device or PBX. What is this IP from the logs Remote host can't match request CANCEL to call '45c0279949faecae6da23a137f782525@94.23.147.9:5060 This looks like a...
  8. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun I see. Ok you could also be having issues with connectivy and high latency becuase you are connecting to another country. There are to many variables that could cause the problem like how good is the connectivity on the PBX in the...
  9. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Just checked your config again. Add the following lines to your config: disallow=all allow=g729 Thius will force it to use G729 else you might be using G711. Check under FreePBX > Settings > SIP settings that g729 is selected and...
  10. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Looks like your stuck between a rock and a hard place. What device actually establishing the VPN tunnel? Are you connecting this device to a provider or your own asterisk PBX somewhere else? If its your own asterisk PBX then I would...
  11. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun I see. Thats terrible what happened to freedom of communication... You are not hidding anything with PPTP and there are huge overheads on your traffic so a tipical G729 call that uses an average of 24kbps up and down will add about...
  12. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Hmmm. What VPN are you using? The only one's that work well with VoIP is OpenVPN and IPSec. PPTP and L2TP is a big NO NO! Why do you need to hide it from your ISP? Do they shape and block VoIP traffic also what type of connectifity...
  13. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun This is what I mean by a good link to your provider: PING my provider ip (my provider ip) 56(84) bytes of data. 64 bytes from my provider ip: icmp_req=1 ttl=51 time=23.9 ms 64 bytes from my provider ip: icmp_req=2 ttl=51 time=3.77 ms...
  14. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun You could increase it to what ever you like with "qualify=yes" and change it to something like "qualify=4000". The default is 2000ms. Even if you ignore the high latency you will still get dropped calls. While in a call go to your...
  15. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Hi, I would say that if your calls disconnected after exactly 20 sec every time then it's because your provider does not have a route back to your SIP device but you mentioned that its random so I looked at your logs and saw right at...
  16. Voice only working one side.(receive voice from SIP trunk)

    Re: Voice only working one side.(receive voice from SIP trun Hey Poision, As mensioned in this forum. Its a common probem and mostly related to the UDP RTP traffic through a firewall. Firstly for security reasons you do not need to open your ports up to the entire world. Keep them closed and...
  17. call recording send by email

    Hi satchid, I just posted a reply to http://www.elastix.org/component/kunena ... calls.html. I modified the script and it works very well. Hope this helps you. See below: I also installed ssmtp and change mutt config to use ssmtp instead of sendmail because its extremely light MTA and I can...
  18. Asterisk Recording deliver "monitored" calls?

    Re: Asterisk Recording deliver Hey Guys. I was looking arround the net for this exact script and found it. I have dusted it off and made some modifications. The script as it is works well for outgoing calls but not incoming. I am using info from the asteriskcdrdb and using some IF conditions...
  19. To Linux Gurus-Limit of # of files in folder-Ext3

    Hey guys! Thanks a mil for all the ideas. I put it to the test and got a very nice system going with auto archiving and sorting the recordings into date folders. Here is what I did if you are interested. 1. Create a script in /var/lib/asterisk/bin/ called archiver 2. Important to chmod 777...
  20. Directed pickup?

    Not to worry I figured out my problem. It works 100% when you assign a DID to the extension so my advise when setting up a system is this: Setup extensions with DID's then make sure you assign a call group and pickup group in the extensions. Add your follow me's as you like and directed...
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