asterisk users - Sep 2005 - Re: Polycom randomly fails outbound calls,

Hi Andres -
> I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301
>
> The Polycom misses 1 out of 2 dialout calls, this is the full log  
> from a
> call which didn't go through.
>
>  303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on
> 'f4e376c3-7531ff39-c86f6812@192.168.1.18' of Response 2: Found
>  303095 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown
> to ulaw
>  303096 Sep 14 10:45:15 DEBUG[15427]: Didn't get a frame from channel:
> SIP/pstn_2-1f35
>  303097 Sep 14 10:45:15 DEBUG[15427]: Bridge stops bridging channels
> SIP/200-0db1 and SIP/pstn_2-1f35
>
> I guess the line 303096 is the more relevant, but I don't know  
> where to
> start troubleshooting it.
Line 303095 is probably relevant, too.  What codec is the phone  
configured to try first?  It looks like the phone is trying to use  
something asterisk doesn't understand, or is not configured for.   
Maybe set the phone to ulaw instead.

Also, what dtmfmode are you using?  Can we look at your sip.conf from  
asterisk, and the config files for your Polycom phone?

- Noah

Noah Miller wrote:
> Hi Andres -
> The two that we have are just used as lobby phones.  They're good  
> little phones, but if you have the money, I'd definitely recommend  
> the IP501 instead.  The screen is MUCH better, and having full  
> speakerphone is great!  Plus the 500/501 just feels a little more solid.
Yeah, I think it was a wrong move going for the 301 instead,
>
>
> Hmm.  I'm not sure either.  I've never used AMP before (except for
a
> quick glance at asterisk@home).  If you can change the sip settings,  
> I don't think it should matter.
I am not using A@H, it didn't work well for me, I just using AMP over 
asterisk, and yes, sip are 100% tweakable,
how do you configure your system, all by hand?
>
> Well, the two weird things I see here are the type setting and the  
> host.  Type is set to peer, but there doesn't seem to be a  
> corresponding user definition (AFAIK, all peers have to have users).   
> You might try changing it to "type=friend" instead (like 201).
I did it, it was set to peer just because I red somewhere that Polys 
didn't like friend type,
>
> For the host setting, this is the address of the sip device, and not  
> the asterisk server.  If you have the Polycom set to a static address  
> of 192.168.1.18, all is well.  If your Polycom is set to DHCP (this  
> is the default), you should use "host=dynamic"
it's fixed to 18
>
> A couple of things that I know you don't need:
> nat=never
> qualify=no
I took them off too, I got them from the only "how to" I found about
amp
and the polycom,
>
>
> - Noah
>
>
Thanks ,
I hope I can help you same day,

Andres