[asterisk-users] inbound sip rtcp hangup

[Vincent Regnard]

Hi all,

I have configured a connection my sip voip provider. I can make outbound 
call without trouble. But I cannot recieve voip calls. The sip 
negociation seams to start well but at some point during the rtcp 
dialog, things seems to block.

As you can see on the above log sample, I recieve some rtcp packets as 
long I can ear the ring tone on my soft phone (a tcpdump confirms the 
reception only, nothing goes out). But no sip channel for my voip 
provider ever starts on my * server. Is there anything you would suggest 
I could make or test to change this beahviour ? Is there something in 
this log that catches your attention more than me ?

Thanks for your help, comments and suggestions.
Vincent

******

192.168.0.30 is the LAN interface of my asterisk server,
62.147.170.96 its public IP. freephonie.net is my voip provider.


Jul 19 19:41:00 VERBOSE[23199] logger.c:     -- Zap/2-1 answered 
SIP/220-081c4618
Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel SIP/220-081c4618 to 
read format slin
Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel Zap/2-1 to write 
format slin
Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel Zap/2-1 to read 
format slin
Jul 19 19:41:00 DEBUG[23199] channel.c: Set channel SIP/220-081c4618 to 
write format slin
Jul 19 19:41:00 DEBUG[23199] chan_sip.c: sip_answer(SIP/220-081c4618)
Jul 19 19:41:00 DEBUG[15342] devicestate.c: Changing state for Zap/2 - 
state 2 (In use)
Jul 19 19:41:00 DEBUG[15342] chan_sip.c: Checking device state for peer 220
Jul 19 19:41:00 DEBUG[15342] devicestate.c: Changing state for SIP/220 - 
state 2 (In use)
Jul 19 19:41:00 DEBUG[15342] chan_iax2.c: Checking device state for 
device 200
Jul 19 19:41:00 DEBUG[15342] chan_iax2.c: iax2_devicestate: Found peer. 
What's device state of 200? addr=0, defaddr=0 maxms=0, lastms=0
Jul 19 19:41:00 DEBUG[15342] chan_sip.c: Checking device state for peer 220
Jul 19 19:41:00 DEBUG[23205] app_queue.c: Device 'Zap/2' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
Jul 19 19:41:00 DEBUG[23206] app_queue.c: Device 'SIP/220' changed to 
state '2' (In use) but we don't care because they're not a member of any 
queue.
Jul 19 19:41:00 DEBUG[23199] rtp.c: Ooh, format changed from unknown to alaw
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = No match Their Call ID: 
xnrpwvawplnvjof at 192.168.1.160 Their Tag gxqny Our tag: as324b2206
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for 
01-00561-0180cd86-4232bf2a1 at freephonie.net - INVITE (With RTP)
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: **** Received INVITE (5) - 
Command in SIP INVITE
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Setting NAT on RTP to 0
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = No match Their Call ID: 
01-00561-0180cd86-4232bf2a1 at freephonie.net Their Tag 
01-00561-0180cd87-27cdfef30 Our tag: as4b0f6c15
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = Found Their Call ID: 
xnrpwvawplnvjof at 192.168.1.160 Their Tag gxqny Our tag: as324b2206
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: **** Received ACK (6) - Command 
in SIP ACK
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Stopping retransmission on 
'xnrpwvawplnvjof at 192.168.1.160' of Response 638: Match Found
Jul 19 19:41:01 DEBUG[23199] rtp.c: Got RTCP report of 52 bytes
Jul 19 19:41:01 DEBUG[15347] chan_sip.c: = Found Their Call ID: 
01-00561-0180cd86-4232bf2a1 at freephonie.net Their Tag 
01-00561-0180cd87-27cdfef30 Our tag: as4b0f6c15
Jul 19 19:41:01 DEBUG[15347] chan_sip.c: **** Received ACK (6) - Command 
in SIP ACK
Jul 19 19:41:01 DEBUG[15347] chan_sip.c: Stopping retransmission on 
'01-00561-0180cd86-4232bf2a1 at freephonie.net' of Response 25491653: Match 
Found
Jul 19 19:41:07 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:11 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for 
(No Call-ID) - OPTIONS (No RTP)
Jul 19 19:41:11 DEBUG[15347] chan_sip.c: = Found Their Call ID: 
262604ac7c68bdcb0f8b009767da7f6c at 192.168.0.30 Their Tag  Our tag: as2e11d2f1
Jul 19 19:41:11 DEBUG[15347] chan_sip.c: Stopping retransmission on 
'262604ac7c68bdcb0f8b009767da7f6c at 192.168.0.30' of Request 102: Match Found
Jul 19 19:41:12 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for 
(No Call-ID) - OPTIONS (No RTP)
Jul 19 19:41:12 DEBUG[15347] chan_sip.c: = Found Their Call ID: 
68d37ecc373ad3c43d6961007abecd8d at 62.147.170.96 Their Tag  Our tag: 
as5cc60643
Jul 19 19:41:12 DEBUG[15347] chan_sip.c: Stopping retransmission on 
'68d37ecc373ad3c43d6961007abecd8d at 62.147.170.96' of Request 102: Match Found
Jul 19 19:41:12 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:15 DEBUG[15347] chan_sip.c: Auto destroying call 
'01-00561-0180cd86-4232bf2a1 at freephonie.net'
Jul 19 19:41:19 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:24 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:31 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:34 DEBUG[23199] rtp.c: Got RTCP report of 96 bytes
Jul 19 19:41:34 DEBUG[15347] chan_sip.c: = Found Their Call ID: 
xnrpwvawplnvjof at 192.168.1.160 Their Tag gxqny Our tag: as324b2206
Jul 19 19:41:34 DEBUG[15347] chan_sip.c: **** Received BYE (8) - Command 
in SIP BYE
Jul 19 19:41:34 DEBUG[23199] channel.c: Didn't get a frame from channel: 
SIP/220-081c4618
Jul 19 19:41:34 DEBUG[23199] channel.c: Bridge stops bridging channels 
SIP/220-081c4618 and Zap/2-1
Jul 19 19:41:34 DEBUG[23199] channel.c: Hanging up channel 'Zap/2-1'


My sip config is as follows (sip register is ok):

[08XXXXXX]
username=08XXXXXX
type=user
canreinvite=no
secret=XXXXXX
qualify=yes
nat=never
permit=212.27.52.5/255.255.255.255
deny=0.0.0.0/0.0.0.0
context=from-pstn
allow=alaw


[freephonie_outbound]
username=08XXXXXX
type=peer
secret=XXXXXX
qualify=yes
host=freephonie.net
fromuser=08XXXXXX
fromdomain=freephonie.net
context=from-internal
nat=never
canreinvite=no
allow=alaw