I have installed Asterisk on Debian etch on a Xen virtual machine of Linode (just a provider of Xen vm). It is working all pretty well, Dahdi_dummy is loading and when calling the conference, the message "you are the first to enter this conference’ is playing. After just hearing a snip of the moh (I used the m parameter) there is no sound while tcpdump -n port not 22 shows that there are a lot of UDP packages are send and received on the server.
04:22:07.198259 IP xx.xxx.119.xx.40444 > xx.xxx.137.xx.14452: UDP, length 172
04:22:07.199701 IP xx.xxx.137.xx.14452 > xx.xxx.119.xx.40444: UDP, length 172
04:22:07.216322 IP xx.xxx.119.xx.40444 > xx.xxx.137.xx.14452: UDP, length 172
04:22:07.219708 IP xx.xxx.137.xx.14452 > xx.xxx.119.xx.40444: UDP, length 172
04:22:07.239690 IP xx.xxx.137.xx.14452 > xx.xxx.119.xx.40444: UDP, length 172
When tcpdump is stopped this is the last output
1745 packets captured
1748 packets received by filter
0 packets dropped by kernel
This is the debug output
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:3787 sip_answer: SIP answering channel: SIP/31201111111-081ea218
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:7002 transmit_response_with_sdp: Setting framing from config on incoming call
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:6725 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:6726 add_sdp: ** Our prefcodec: 0x0 (nothing)
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:6843 add_sdp: – Done with adding codecs to SDP
[Jul 29 04:35:17] DEBUG[5368]: channel.c:2504 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=25)
[Jul 29 04:35:17] DEBUG[5368]: chan_sip.c:6944 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw)
[Jul 29 04:35:17] DEBUG[5368]: pbx.c:1843 pbx_extension_helper: Launching ‘MeetMe’
– Executing [31201111111@inbound:2] MeetMe(“SIP/31201111111-081ea218”, “1234|m”) in new stack
== Parsing ‘/etc/asterisk/meetme.conf’: [Jul 29 04:35:17] DEBUG[5368]: config.c:886 config_text_file_load: Parsing /etc/asterisk/meetme.conf
Found
[Jul 29 04:35:17] DEBUG[5368]: chan_dahdi.c:8391 dahdi_request: Using channel -2
[Jul 29 04:35:17] DEBUG[5368]: channel.c:3085 set_format: Set channel DAHDI/pseudo-2055445543 to read format slin
[Jul 29 04:35:17] DEBUG[5368]: channel.c:3085 set_format: Set channel DAHDI/pseudo-2055445543 to write format slin
– Created MeetMe conference 1023 for conference ‘1234’
[Jul 29 04:35:17] DEBUG[5368]: devicestate.c:302 ast_device_state_changed_literal: Notification of state change to be queued on device/channel meetme:1234
[Jul 29 04:35:17] DEBUG[5368]: channel.c:3085 set_format: Set channel SIP/31201111111-081ea218 to write format gsm
[Jul 29 04:35:17] DEBUG[5368]: rtp.c:2885 ast_rtp_write: Ooh, format changed from unknown to alaw
[Jul 29 04:35:17] DEBUG[5368]: rtp.c:2902 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160
[Jul 29 04:35:17] DEBUG[5368]: channel.c:1875 ast_settimeout: Scheduling timer at 160 sample intervals
– <SIP/31201111111-081ea218> Playing ‘conf-onlyperson’ (language ‘en’)
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 31201111111
[Jul 29 04:35:17] DEBUG[3011]: chan_sip.c:16953 sip_devicestate: Checking device state for peer 31201111111
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:287 do_state_change: Changing state for SIP/31201111111 - state 1 (Not in use)
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 31201111111
[Jul 29 04:35:17] DEBUG[3011]: chan_sip.c:16953 sip_devicestate: Checking device state for peer 31201111111
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:287 do_state_change: Changing state for SIP/31201111111 - state 1 (Not in use)
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:157 ast_device_state: Checking if I can find provider for “meetme” - number: 1234
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:230 getproviderstate: Checking provider SLA with meetme
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:230 getproviderstate: Checking provider Meetme with meetme
[Jul 29 04:35:17] DEBUG[3011]: devicestate.c:287 do_state_change: Changing state for meetme:1234 - state 2 (In use)
[Jul 29 04:35:17] DEBUG[3038]: app_queue.c:675 handle_statechange: Device ‘SIP/31201111111’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jul 29 04:35:17] DEBUG[3038]: app_queue.c:675 handle_statechange: Device ‘SIP/31201111111’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jul 29 04:35:17] DEBUG[3026]: chan_sip.c:4745 find_call: = Found Their Call ID: 1035172009-83CDE46516962907337@xx.xxx.119.xx Their Tag GR52RWG346-34 Our tag: as7e744e42
[Jul 29 04:35:17] DEBUG[3026]: chan_sip.c:16179 handle_request: **** Received ACK (6) - Command in SIP ACK
[Jul 29 04:35:17] DEBUG[3026]: chan_sip.c:2238 __sip_ack: Stopping retransmission on ‘1035172009-83CDE46516962907337@82.146.119.38’ of Response 1: Match Found
[Jul 29 04:35:20] DEBUG[5368]: channel.c:1875 ast_settimeout: Scheduling timer at 0 sample intervals
[Jul 29 04:35:20] DEBUG[5368]: channel.c:1875 ast_settimeout: Scheduling timer at 0 sample intervals
[Jul 29 04:35:20] DEBUG[5368]: channel.c:3085 set_format: Set channel SIP/31201111111-081ea218 to write format alaw
[Jul 29 04:35:20] DEBUG[5368]: channel.c:3085 set_format: Set channel SIP/31201111111-081ea218 to write format slin
[Jul 29 04:35:20] DEBUG[5368]: channel.c:3085 set_format: Set channel SIP/31201111111-081ea218 to read format slin
[Jul 29 04:35:20] DEBUG[5368]: app_meetme.c:1817 conf_run: Placed channel SIP/31201111111-081ea218 in DAHDI conf 1023
[Jul 29 04:35:20] DEBUG[5368]: rtp.c:2712 ast_rtp_raw_write: Difference is 1176, ms is 167
[Jul 29 04:35:22] DEBUG[5368]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes
[Jul 29 04:35:22] DEBUG[3026]: chan_sip.c:2143 __sip_autodestruct: Auto destroying SIP dialog ‘7fb97d4d3098e07d12cd998366bcd02f@127.0.1.1’
[Jul 29 04:35:22] DEBUG[3026]: chan_sip.c:3408 sip_destroy: Destroying SIP dialog 7fb97d4d3098e07d12cd998366bcd02f@127.0.1.1
Really destroying SIP dialog ‘7fb97d4d3098e07d12cd998366bcd02f@127.0.1.1’ Method: REGISTER
[Jul 29 04:35:27] DEBUG[5368]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes
[Jul 29 04:35:32] DEBUG[5368]: rtp.c:923 ast_rtcp_read: Got RTCP report of 64 bytes
ubuntu*CLI>
Disconnected from Asterisk server
What can I do to make Meetme() run on this Debian Etch Linode Xen vm