[Asterisk-Users] Polycom randomly fails outbound calls,

[Andres Paglayan]

Hi All,

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.

 303091 Sep 14 10:45:15 VERBOSE[15427]:     -- SIP/pstn_2-1f35 answered 
SIP/200-0db1
 303092 Sep 14 10:45:15 VERBOSE[15427]:     -- Attempting native bridge 
of SIP/200-0db1 and SIP/pstn_2-1f35
 303093 Sep 14 10:45:15 DEBUG[15427]: Ooh, format changed from unknown 
to ulaw
 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on 
'f4e376c3-7531ff39-c86f6812 at 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
 303098 Sep 14 10:45:15 DEBUG[15427]: update_user_counter(ww4902758) - 
decrement outUse counter
 303099 Sep 14 10:45:15 DEBUG[15427]: ww4902758 is not a local user
 303100 Sep 14 10:45:15 DEBUG[15427]: Exiting with DIALSTATUS=ANSWER.
 303101 Sep 14 10:45:15 VERBOSE[15427]:   == Spawn extension 
(macro-dialout-trunk, s, 17) exited non-zero on 'SIP/200-0db1' in macro 
'        dialout-trunk'
 303102 Sep 14 10:45:15 VERBOSE[15427]:   == Spawn extension 
(from-internal, 4902758, 1) exited non-zero on 'SIP/200-0db1'


The Poly dials out using the SPA3000 FXO, all other phones connect to 
SPA300 FXO from SPA2000 FXS and they work fine when dialing out,

What I noticed is that in the successful calls you could hear the tones 
going out, in the calls that fail there's only silence.

I added two ww to check if it was a timing issue before getting tones, 
but is not.

I guess the line 303096 is the more relevant, but I don't know where to 
start troubleshooting it.

Any clue or tip will be appreciated,

Thank you,