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hfourie

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About hfourie

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  1. That is the article I used to come up with the current layout. Something is definetely different when using the Polycom. Just tested X-Lite again. Works fine. Even with SIP replacement and IP routing fields used. POLYCOM!
  2. ..that did not work. internal calls are fine, but dialing out there is no audio. Either direction. So i put the SIP replacement field and the other one back, now the polycom works internal, but still the same dial out problem. Again, Snom works fine. Your network sugestion, that is basically what I have. One interface has a 1 to 1 NAT setup, which is for external access and SIP Provider connection, and the other interface is for internal only. Dont quite know where to go with this next, like I said before, Snom 360 works fine. btw, is my IP Routing List syntax correct?
  3. I think it is working now.... Under Network, RTP Port Settings, Filter By IP Address is enabled by default. changed to disabled...and put my sip replacement and the other one back into pbxnsip. Seems to work. Well audio now def works internally. Will test some connections from outside and post results. Thanks for the help. Your RTP suggestion was spot on.
  4. This is my SIP Replacement List: 10.1.1.10/external_IP IP Routing List: 10.1.1.0/255.255.255.0/10.1.1.6 0.0.0.0/0.0.0.0/external_IP my server, CentOS has 2 adapters. 10.1.1.6 does nota have a gateway configured, so it is used for internal calls and the internal phones connect to it. 10.1.1.10 has a def gw, and it is natted to the internet for users connecting from outside. Do I have something wrong on the 2 fields?
  5. Hi, I am setting up my first Polycom phone with pbxnsip. Have used Snom before. On pbxnsip I am using both the SIP IP Replacement List and IP Routing List. This enabled good calls both internally and calls from clients connected from the internet using xLite. Ive setup the Polycom 330 and is currently logged into pbxnsip. When making calls though, no audio in either direction is heard. When I remove both lines (SIP IP Replacement List and IP Routing List) audio is restored, but only one way with outgoing audio working. I need the tow line (SIP IP Replacement List and IP Routing List) as this resolved some issues I had where when hanging up a call did not terminate the other side etc. And it works fine with the Snom. Am I missing a setting that needs to be added to the Polycom? Many Thanks, any help appreciated.
  6. Hi all, Come across a really anoying problem. Seems like since I enabled the SIP Replacement field (to enable users to plug snom on internet and make calls) if someone from outside phones in, and then hang up, the phone on pbxnsip still keeps ringing. If I dont pick it up, it goes to voicemail, and I get lots of silent voicemails. The only way to get fix it, is to remove the SIP Replacement field, reboot server (Windows).... put info back, reboot. It then works fine for few haours, maybe a day. Then same problem. Help would be appreciated.
  7. Hi, Dont know if this is a pbxnsip issue, or a eyeBeam issue. When a call is recieved, using eyebeam, via the Auto attendant, caller selects the extension, then on the recieving side, in eyebeam i see the callers number. Great, this is how it should be. ... however... When a caller comes via a Agent group or Hunt Group, the eyebeam interface shows telno@ipaddress. The ip address is the IP of the SIP provider. This causes problems when wanting to return the call.. Anyone have ideas. Also using a snom360, which doesnt do this, but why is it ok on AA transfers, and not Hunt Group? Help apprecated.
  8. Thanks I'm gonna try it now. btw its www.kapanga.net for anyone else wanting to try it...
  9. Do you know if they still make it, cant find it on their website. Do you know of an alternatuve? would be nice if you could use Windows Mobile device to make local calls through pbxnsip!!
  10. Hi, Does anyone have a VOIP Client recommendation for Windows Mobile devices with WMD 5/6. ... and have it working correctly? Thanks
  11. Hi there, This is exactly what I am trying to do, but not having any luck. Currently have a demo license, and as 14 days is not much, I fear time will run out before I get to properly evaluate the product. Im trying to use the Exchange AA and OVA etc through a SIP provider via pbxnsip. My SIP Trunk is running, but not getting the exchange hookup working properly. More detail on your config will be much appreciated. Many thanks
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