[Asterisk-Users] Re: Polycom randomly fails outbound calls,

[Andres Paglayan]

More information in this thread,

This Poly 301 sometimes rings out sometimes doesn't,
it calls out through * using spa3000's fxo,

I got this log

    1129 Sep 14 15:03:55 DEBUG[15702]: Outgoing Call for ww9821642
    1130 Sep 14 15:03:55 DEBUG[15702]: ww9821642 is not a local user
    1131 Sep 14 15:03:55 VERBOSE[15702]:     -- Called pstn_2/ww9821642
    1132 Sep 14 15:03:55 DEBUG[15690]: Acked pending invite 102
    1133 Sep 14 15:03:55 DEBUG[15690]: Stopping retransmission on 
'1cd1aa2160ed236e5e5db09e2e7916d8 at 192.168.1.8' of Request 102: Found
    1134 Sep 14 15:03:55 DEBUG[15690]: (Provisional) Stopping 
retransmission (but retaining packet) on 
'1cd1aa2160ed236e5e5db09e2e7916d        8 at 192.168.1.8' Request 103: Found
    1135 Sep 14 15:03:55 DEBUG[15690]: (Provisional) Stopping 
retransmission (but retaining packet) on 
'1cd1aa2160ed236e5e5db09e2e7916d        8 at 192.168.1.8' Request 103: Found
    1136 Sep 14 15:03:55 VERBOSE[15702]:     -- SIP/pstn_2-4fc7 is ringing
    1137 Sep 14 15:03:56 DEBUG[15690]: Acked pending invite 103
    1138 Sep 14 15:03:56 DEBUG[15690]: Stopping retransmission on 
'1cd1aa2160ed236e5e5db09e2e7916d8 at 192.168.1.8' of Request 103: Found
    1139 Sep 14 15:03:56 DEBUG[15690]: build_route: Contact hop: pstn 
spa 3002 <sip:ww9821642 at 192.168.1.17:5061>
    1140 Sep 14 15:03:56 VERBOSE[15702]:     -- SIP/pstn_2-4fc7 answered 
SIP/200-eb9f
    1141 Sep 14 15:03:56 VERBOSE[15702]:     -- Attempting native bridge 
of SIP/200-eb9f and SIP/pstn_2-4fc7
    1142 Sep 14 15:03:56 DEBUG[15702]: Ooh, format changed from unknown 
to ulaw
    1143 Sep 14 15:03:56 DEBUG[15690]: Stopping retransmission on 
'33aa45dd-b2e2caf3-7ef5c3c at 192.168.1.18' of Response 2: Found
    1144 Sep 14 15:03:56 DEBUG[15702]: Ooh, format changed from unknown 
to ulaw
    1145 Sep 14 15:04:02 DEBUG[15690]: Auto destroying call 
'5d3a35b8-f15db87f at 192.168.1.16'
    1146 Sep 14 15:04:02 DEBUG[15690]: Auto destroying call 
'b1233c64-836281c3 at 192.168.1.16'
    1147 Sep 14 15:04:23 DEBUG[15702]: Didn't get a frame from channel: 
SIP/200-eb9f
    1148 Sep 14 15:04:23 DEBUG[15702]: Bridge stops bridging channels 
SIP/200-eb9f and SIP/pstn_2-4fc7
    1149 Sep 14 15:04:23 DEBUG[15702]: update_user_counter(ww9821642) - 
decrement outUse counter
    1150 Sep 14 15:04:23 DEBUG[15702]: ww9821642 is not a local user
    1151 Sep 14 15:04:23 DEBUG[15702]: Exiting with DIALSTATUS=ANSWER.
    1152 Sep 14 15:04:23 VERBOSE[15702]:   == Spawn extension 
(macro-dialout-trunk, s, 17) exited non-zero on 'SIP/200-eb9f' in macro 
'        dialout-trunk'
    1153 Sep 14 15:04:23 VERBOSE[15702]:   == Spawn extension 
(from-internal, 9821642, 1) exited non-zero on 'SIP/200-eb9f'
    1154 Sep 14 15:04:23 VERBOSE[15702]:     -- Executing 
Macro("SIP/200-eb9f", "hangupcall") in new stack
    1155 Sep 14 15:04:23 VERBOSE[15702]:     -- Executing 
ResetCDR("SIP/200-eb9f", "w") in new stack
    1156 Sep 14 15:04:23 VERBOSE[15702]:     -- Executing 
NoCDR("SIP/200-eb9f", "") in new stack


I am forcing all, the sipura and the polycom to use only ulaw, but still 
I get that 'unknown' protocol, and no frames, nor from the polycom, nor 
from the spa

Also this happends randomly there's nothing that gives a pattern,


I am using rfc2833 as dtmf mode

I already tweaked the dialplan.digitmap="" (to an empty string) so
everything gets out.

my phone's sip.cfg codec setting looks like
<preferences voice.codecPref.G711Mu="1" voice.codecPref.G711A="2"
voice.codecPref.G729AB="3" voice.codecPref.IP_4000.        G711Mu="1"
voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/>

And ulaw is set as preferred in the extension.

as shown in sip.conf

[general]

  port = 5060           ; Port to bind to (SIP is 5060)
  bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
  disallow=all
  allow=ulaw
  allow=alaw
  context = from-sip-external ; Send unknown SIP callers to this context
  callerid = Unknown


Thanks again,