[asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls

Dan Serban dserban at lodgingcompany.com
Thu Sep 7 07:51:51 MST 2006


Alberto Sagredo wrote:
> I use canreinvite=yes in my config files, and it does work, so maybe its
> a spa 941 misconfiguration.
> 
> I think if nat=no sometime it has problems if you are behind NAT, but
> under same network it must not fail.
> 

I am behind a NAT, though the whole network is seperate, ie, I don't
proxy through a NAT, the machiene is fully internal.  Like I had
described in my previous message, you could forward calls to other
phones, though if you weren't quick enough the chance of you hearing the
caller were slim.

> Which firmware are you running on spas?
> 

4.1.15

And BTW, I'm not exactly happy with these phones, though compared with
what's offered out there, I'm sure we've made a wise decision.  There's
still a nagging feeling that the Polycom's would have done a better job.

> Dan Serban escribió:
>> Alberto Sagredo wrote:
>>  
>>> I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works
>>> fine. Are you canreinvite=yes ?.
>>>
>>> I have not been notice any problem related to transferring calls (blind
>>> and attended)
>>>
>>>     
>>
>> Thank you for your response, it gave me a nudge to check the
>> configuration in the sip.conf file.  It seems that if I set
>> canreinvite=no for every SIP peer, it works!
>>
>> And I have found no other adverse effects.  Strange issue...
>>
>>  
>>> Regards
>>>
>>> Dan Serban escribió:
>>>    
>>>> I have a system running Asterisk 1.2.10 (Debian packaging) and about 50
>>>> Linksys SPA-942 phones, after the initial config and mass deployment of
>>>> the phones everything looks like it's configured well.
>>>>
>>>> When an incoming call is answered and then attempted to be "xfer'ed"
>>>> via
>>>> the soft button on the phone itself, it seems that if you hit the
>>>> button
>>>> twice in quick succession, there is no problem (effectively a blind
>>>> transfer), if then I try to tell the other extension that "Joe is
>>>> calling to sell you a fridge" and hit xfer, the calling party cannot
>>>> hear what that person at the extension is saying.  Sometimes the tables
>>>> are fully turned, the caller can hear, but the operator can't hear a
>>>> thing.
>>>>
>>>> One thing's for sure, if you hit the button quickly (blind transfer) it
>>>> works no problem at all.
>>>>
>>>> This is what I see asterisk saying when I transfer the call
>>>> unsuccessfully.
>>>>
>>>> == Spawn extension (macro-stdexten, s, 1) exited non-zero on
>>>> 'SIP/82-006d42a0<ZOMBIE>' in macro 'stdexten'
>>>> == Spawn extension (macro-stdexten, s, 1) exited non-zero on
>>>> 'SIP/82-006d42a0<ZOMBIE>'
>>>>
>>>> I've looked at the macro with a fine tooth comb, I cannot see any
>>>> problems with it whatsoever, (though that doesn't mean that my
>>>> ignorance
>>>> isn't getting in the way).
>>>>
>>>> I found some mention on the digium mantis bug tracker, here's the link:
>>>>
>>>> http://bugs.digium.com/view.php?id=7421
>>>>
>>>> Before I try and patch the source (which I'm hesitant to do since I run
>>>> the debian packages), is there another solution or maybe an
>>>> unidentified
>>>> issue that I haven't been able to decipher?
>>>>
>>>> If there's more information that I can provide to solve this problem,
>>>> I'd be happy to do so.
>>>>
>>>> Thank you.
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