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330 no audio


hfourie
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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.

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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?

 

I believe that the replacement list is not 100 % clean. Probably the snom is not so picky with the Contact headers like the Polycom, and the conversation works also with this flaw. You will see if the PBX SIP messages contain a Contact header that can be reached from the phones.

 

The other thing that would come to my mind would be the RTP port range which could be different between snom and Polycom.

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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?

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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.

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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.

 

:)

 

If you have a public IP address and the possibility to put the PBX into a DMZ (transparent, no NAT) then such a setup is a lot easier. You can just configure one NIC for this public interface and the other NIC with a private IP address.

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..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?

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..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?

 

What router are you using? Does it support transparent DMZ? If that is the case you can pretty much use http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.

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What router are you using? Does it support transparent DMZ? If that is the case you can pretty much use http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses.

 

 

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! :(

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