Question about Avaya 4610SW IP Phone

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IP phone to IP phone through an IP trunk Issue

Are there any ideas... when an IP phone calls another IP phone through an IP trunk, one end cannot hear the other. The other IP phone would be able to hear just fine.

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Scenario 1 : IP Trunk is created between two Avaya S8xx servers

Solution: Near site's IP phone's IP address must be pingable from the Far site's CLAN & MEDPRO cards (those which have been used to create the trunk). If the above doesn't happen, then necessary network changes must be made to ensure reachability from one side's CLAN/MEDPRO to the far side IP phones.

Scenario 2 : IP Trunk is created between an Avaya S8xx server and IP Office

Solution: Proper IP route must be created in IP Office to accept the packets coming from the IP phones. An entry for their network address must be made in the IP Office

Posted on Apr 02, 2009

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Is this an IP office? If so, maybe try to untick the "direct media path" box in the extension tab.

Posted on Jul 08, 2008

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1 Answer

How to fix snom 300 because it cannot received calls


Is your phone connecting to a hosted PBX provider or is it connecting to a premise-based IP PBX? Incoming and outgoing calls are usually handled by two separate connections between your PBX and the service provider SIP trunk. Sounds like a registration problem with the SIP trunk. There is nothing you can do from the phone itself to fix this.

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I use polycom Sound Point IP 550 phone. For an incoming or outgoing call after some duration the other end of the caller is not able to listen to me. However I can listen to them, once understanding that...


This sounds like an RTP stream issue. While SIP runs on port 5060 by default, the actual audio is run on much higher ports, 10000 and up. If your pbx system is not on a public address and you are forwarding traffic to your pbx via a router, check the RTP port range in your pbx and make sure you are forwarding those ports correctly. By putting the call on hold, you stop sending out audio traffic restart sending audio when you pick the call back up. By forwarding the entire RTP range to your pbx, this should fix your issue.

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Set it up just like you would dial it. example 917145551234. You have to do this from your extn login and will have to have trunk to trunk enabled under your class of service.

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The mute button is the top right button, it's a clear colored button. When it's lit and orange it's muted.

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Upgrade to the new skype version!!

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I gather you are using an IP phone. I would have to say that the only way to trouble shoot is to use a protocol analyzer. If you connect it within the IP path you will be able to see both ends of the call and determine if the call was received or not and if the other end is responding to your call. Your IP PBX might be the problem I would run a diagnostic on it and if it fails, I would suggest calling the manufacture to see if there are any tests you can run.

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2 Answers

For incoming calls, the caller cannot hear me for 2-10 seconds after I pick up


You've got an IP issue somewhere in your setup. It sounds like the call is being lost and then the phone is querying the DNS server to find out where it should be looking for the call. Do you have a VOIP server name or it's IP address in the setup? Are you using DHCP or manually assigning IP addresses? If the VOIP server is not on a static IP, I would fix that first. Post back here the answers and we can look at it further.

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