Hi there Ryan !

What about the asterisk side, what is the termination? Analog, IP trunk, PRI ?

Do you have this problem doing extension to extension calls?

---
Andre Courchesne - Consultant
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Skype: VoipForces



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On 2011-05-13, at 8:21 PM, Henry Coleman wrote:

> Hi Ryan, it sounds like a local problem. As Lloyd suggests a SIP phone at
> another location would soon uncover where the problem is.
> If you are using a DSL line that is also being used as an analog line you
> should check that the low pass filter is installed for any analog devices
> like a fax machine. You should also check that the local network is wired
> correctly.
> If you set up a SIP extension and send me the details off list (Domain name/
> IP address, extension number and password)  I can check this out for you.
> 
> Henry
> 
> 
> 
> On Fri, May 13, 2011 at 6:31 PM, Aloysius Lloyd 
> <[email protected]>wrote:
> 
>> Hello Ryan,
>> 
>> There are so many factors here
>> 
>> 1. Can you try to connect remote Asterisk server using xlite. What is the
>> results?
>> 
>> 2. Aastra Phone Enable the rport on the network page?
>> 
>> 3. What is the model of the Aastra phone and firmware?
>> 
>> Thanks
>> Lloyd
>> 
>> 
>> On Fri, May 13, 2011 at 4:40 PM, <[email protected]> wrote:
>> 
>>> OK, this one has me stumped.
>>> 
>>> 
>>> 
>>> A customer of ours has a client with Aastra phones connecting to a remote
>>> asterisk
>>> 
>>> 1.6 server. The 4 phones are behind a Linksys wrt160n. This is the
>> problem:
>>> 
>>> 
>>> 
>>> When a public caller calls the aastra phone, after a few minutes the
>> caller
>>> starts to hear ringing in his handset, as if he was calling someone else
>> but
>>> he is still on the same call. Our client does not hear the ringing. Only
>> the
>>> person calling in. Its not consistent ringing either. It will ring once,
>>> stop for 10 seconds, ring twice, stop for 5 seconds and so on. Of course
>> it
>>> becomes impossible to hold a conversation.
>>> 
>>> 
>>> 
>>> The phones are configured to use a STUN server for nat traversal, the sip
>>> ports are staggered so as not to create a conflict (phone 1 = 5060, phone
>> 2
>>> = 5062...)
>>> 
>>> 
>>> 
>>> We tried unplugging all other phones so that there is only one Aastra
>>> connected to the network and the problem continues. I can't find anyone
>>> complaining about anything similar on the net either
>>> 
>>> 
>>> 
>>> Do you have any ideas?
>>> 
>>> 
>>> 
>>> 
>>> 
>>> Ryan Mayoff
>>> Vice-President
>>> 
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>>> 9740 Transcanada Highway
>>> St-Laurent,QC, H4S 1V9
>>> 
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>>> fax:       514-342-9222
>>> 
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> 
> 
> 
> -- 
> * Henry Coleman*
> [image: Picture]

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