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:
[EMAIL PROTECTED] Their Tag gxqny Our tag: as324b2206
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: Allocating new SIP dialog for
[EMAIL PROTECTED] - 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:
[EMAIL PROTECTED] Their Tag
01-00561-0180cd87-27cdfef30 Our tag: as4b0f6c15
Jul 19 19:41:00 DEBUG[15347] chan_sip.c: = Found Their Call ID:
[EMAIL PROTECTED] 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
'[EMAIL PROTECTED]' 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:
[EMAIL PROTECTED] 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
'[EMAIL PROTECTED]' 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:
[EMAIL PROTECTED] Their Tag Our tag: as2e11d2f1
Jul 19 19:41:11 DEBUG[15347] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' 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:
[EMAIL PROTECTED] Their Tag Our tag:
as5cc60643
Jul 19 19:41:12 DEBUG[15347] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' 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
'[EMAIL PROTECTED]'
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:
[EMAIL PROTECTED] 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
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