[Asterisk-Users] MOH & Vegastream

[Issac Simchayof]

When ever we get a call through our VegaStream 50 FXO the MOH for that call
gets turned off. Anyway to troubleshoot this by looking at the log below?


Thanks

Issac


Jun 13 16:00:24 DEBUG[2443] chan_sip.c: Stopping retransmission on
'e1481b63-2d3a91a5-992f4cd6 at 209.219.90.161' of Response 2: Match Found
Jun 13 16:00:25 DEBUG[3107] rtp.c: Ooh, format changed from unknown to ulaw
Jun 13 16:00:26 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:29 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:34 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:36 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:39 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:43 DEBUG[2474] chan_iax2.c: Allocate call number
Jun 13 16:00:43 DEBUG[2474] chan_iax2.c: Registration created on call 1
Jun 13 16:00:43 NOTICE[2474] chan_iax2.c: Registration of 'issacs' rejected:
'Registration Refused' from: '207.174.202.4'
Jun 13 16:00:44 DEBUG[2474] chan_iax2.c: Allocate call number
Jun 13 16:00:44 DEBUG[2474] chan_iax2.c: Registration created on call 2
Jun 13 16:00:44 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:46 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:49 DEBUG[2443] chan_sip.c: Allocating new SIP dialog for
f0fa8f9f-b8016959-15df7004 at 192.168.1.51 - REGISTER (No RTP)
Jun 13 16:00:49 DEBUG[3113] app_queue.c: Device 'SIP/401' changed to state
'1' (Not in use) but we don't care because they're not a member of any
queue.
Jun 13 16:00:49 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:53 DEBUG[2443] chan_sip.c: Allocating new SIP dialog for (No
Call-ID) - NOTIFY (No RTP)
Jun 13 16:00:53 DEBUG[2443] chan_sip.c: Stopping retransmission on
'6f135b5e2fe693884b4f64a509caf861 at 209.219.90.167' of Request 102: Match
Found
Jun 13 16:00:54 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:00:56 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:00:59 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:04 DEBUG[2443] chan_sip.c: Auto destroying call
'f0fa8f9f-b8016959-15df7004 at 192.168.1.51'
Jun 13 16:01:04 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:06 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:01:09 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes	
Jun 13 16:01:14 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:16 DEBUG[3107] rtp.c: Got RTCP report of 80 bytes
Jun 13 16:01:19 DEBUG[3107] rtp.c: Got RTCP report of 76 bytes
Jun 13 16:01:20 VERBOSE[2443] logger.c: -- Started music on hold, class
'default', on SIP/Vega-2127681168-4afe
Jun 13 16:01:20 DEBUG[2443] chan_sip.c: Stopping retransmission on
'e1481b63-2d3a91a5-992f4cd6 at 209.219.90.161' of Response 3: Match Found
Jun 13 16:01:24 DEBUG[3107] rtp.c: Got RTCP report of 56 bytes
Jun 13 16:01:28 VERBOSE[2443] logger.c: -- Stopped music on hold on
SIP/Vega-2127681168-4afe
Jun 13 16:01:28 DEBUG[2443] channel.c: Set channel SIP/Vega-2127681168-4afe
to write format ulaw