It seems to me "nat=yes" is not working correctly in asterisk 1.8.5
rtp set debug on
shows:
Got RTP packet from 10.0.0.110:6000 (type 00, seq 029667, ts 2129095321,
len 000160)
Sent RTP packet to 10.0.0.110:6010 (type 00, seq 065112, ts 2129095320,
len 000160)
I've tried 'nat=yes' 'nat=comedia' it makes no differece.
--
Joseph
On 09/05/11 15:00, Joseph wrote:
I have DID, it registers OK with the provider, but when I try to call this
number (it suppose to ring my Asterisk) asterisk 1.8 does not respond.
sip show peers
Name/username Host Dyn Forcerport ACL Port Status
actio-out/48746612254 81.15.150.20 N 5060 OK (201ms)
sip.conf part:
[general]
context=default
allowguest=no allowoverlap=no
udpbindaddr=0.0.0.0
useragent = Centrala
[actio-out]
type=friend
secret=xxxxxxxx
user=48746612254
username=48746612254
fromuser=48746612254
authname=48746612254
callerpage=48746612254
fromdomain=sip.actio.pl
host=sip.actio.pl
insecure=port,invite
nat=yes
qualify=yes
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
context=from_poland
canreinvite=no
The setting above worked OK with Asteriks 1.4
Here is debug info, which I don't know how to interpret.
-- Executing [901148746612254@internal:1] Dial("SIP/11-00000002",
"SIP/901148746612254@pstn-1270,60,tr") in new stack
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:25695 sip_request_call: Asked to
create a SIP channel with formats: 0x4 (ulaw)
== Using UDPTL CoS mark 5
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:7496 sip_alloc: Allocating new SIP
dialog for [email protected]:0 - INVITE (No RTP)
[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:347 ast_rtp_instance_new: Using
engine 'asterisk' for RTP instance '0x88c3b10'
[Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:474 ast_rtp_new: Allocated
port 16690 for RTP instance '0x88c3b10'
[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:356 ast_rtp_instance_new: RTP
instance '0x88c3b10' is setup and ready to go
[Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:2372 ast_rtp_prop_set: Setup
RTCP on RTP instance '0x88c3b10'
== Using SIP RTP CoS mark 5
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4928 do_setnat: Setting NAT on RTP
to Off
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4936 do_setnat: Setting NAT on UDPTL
to Off
[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1459
ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of
'SIP/pstn-1270-00000003' with that of
'SIP/11-00000002'
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable DIALEDTIME.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable ANSWEREDTIME.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable DIALEDPEERNAME.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable DIALEDPEERNUMBER.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable DIALSTATUS.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable SIPCALLID.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable SIPDOMAIN.
[Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables:
Not copying variable SIPURI.
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:5463 sip_call: Outgoing Call for
901148746612254
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10989 add_sdp: ** Our capability:
0xc (ulaw|alaw) Video flag: False Text flag: False
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10990 add_sdp: ** Our prefcodec: 0x4
(ulaw)
[Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:3054 initialize_initreq:
Initializing initreq for method INVITE - callid
[email protected]:5060
-- Called SIP/901148746612254@pstn-1270
[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'[email protected]:5060' Request 102: Found
[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'[email protected]:5060' Request 102: Found
[Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:538
ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on
0xb6199490
[Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:538
ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on
0xb6199490
[Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:641
ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb6199490
[Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:641
ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb6199490
[Sep 5 14:04:35] DEBUG[26083]: res_rtp_asterisk.c:2393
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x88c3b10'
-- SIP/pstn-1270-00000003 is making progress passing it to SIP/11-00000002
[Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1542
ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/11-00000002'
with that of
'SIP/pstn-1270-00000003'
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1241 ast_rtp_write: Ooh,
format changed from unknown to ulaw
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1272 ast_rtp_write: Created
smoother: format: ulaw ms: 20 len: 160
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1142 ast_rtp_raw_write:
Starting RTCP transmission on RTP instance '0x885bf68'
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 44 bytes
[Sep 5 14:04:39] DEBUG[26083]: chan_sip.c:3974 __sip_ack: Acked pending invite
102
[Sep 5 14:04:39] DEBUG[26083]: chan_sip.c:4012 __sip_ack: Stopping
retransmission on '[email protected]:5060' of Request
102:
Match Found
-- SIP/pstn-1270-00000003 answered SIP/11-00000002
[Sep 5 14:04:39] DEBUG[26209]: chan_sip.c:6297 sip_answer: SIP answering
channel: SIP/11-00000002
[Sep 5 14:04:39] DEBUG[26209]: chan_sip.c:11343 transmit_response_with_sdp:
Setting framing from config on incoming call
[Sep 5 14:04:39] DEBUG[26209]: chan_sip.c:10989 add_sdp: ** Our capability:
0xc (ulaw|alaw) Video flag: True Text flag: True
[Sep 5 14:04:39] DEBUG[26209]: chan_sip.c:10990 add_sdp: ** Our prefcodec: 0x0
(nothing)
[Sep 5 14:04:39] DEBUG[26209]: features.c:3394 clear_dialed_interfaces:
Removing dialed interfaces datastore on SIP/pstn-1270-00000003 since we're
bridging
[Sep 5 14:04:39] DEBUG[26083]: chan_sip.c:4012 __sip_ack: Stopping
retransmission on '[email protected]' of Response 2: Match Found
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1241 ast_rtp_write: Ooh,
format changed from unknown to ulaw
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1272 ast_rtp_write: Created
smoother: format: ulaw ms: 20 len: 160
[Sep 5 14:04:39] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:43] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:43] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:46] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:50] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:51] DEBUG[26209]: res_rtp_asterisk.c:1675 ast_rtcp_read: Got RTCP
report of 68 bytes
[Sep 5 14:04:53] DEBUG[26083]: res_rtp_asterisk.c:2393
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x88c3b10'
[Sep 5 14:04:53] DEBUG[26209]: channel.c:6925 ast_generic_bridge: Didn't get a
frame from channel: SIP/pstn-1270-00000003
[Sep 5 14:04:53] DEBUG[26209]: channel.c:7383 ast_channel_bridge: Bridge stops
bridging channels SIP/11-00000002 and SIP/pstn-1270-00000003
[Sep 5 14:04:53] DEBUG[26209]: res_config_sqlite.c:833 cdr_handler: SQL query:
INSERT INTO ast_cdr
(clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid)
VALUES ('"Joseph"
<11>','11','901148746612254','internal','SIP/11-00000002','SIP/pstn-1270-00000003','Dial','SIP/901148746612254@pstn-1270,60,tr','2011-09-05
14:04:35','2011-09-05 14:04:39','2011-09-05
14:04:53','18','14','ANSWERED','DOCUMENTATION','1315253075.2')
[Sep 5 14:04:53] DEBUG[26209]: channel.c:2807 ast_hangup: Hanging up channel
'SIP/pstn-1270-00000003'
[Sep 5 14:04:53] DEBUG[26209]: chan_sip.c:6096 sip_hangup: Hangup call
SIP/pstn-1270-00000003, SIP callid
[email protected]:5060
[Sep 5 14:04:53] DEBUG[26209]: res_rtp_asterisk.c:2393
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x88c3b10'
[Sep 5 14:04:53] DEBUG[26209]: app_dial.c:2884 dial_exec_full: Exiting with
DIALSTATUS=ANSWER.
[Sep 5 14:04:53] DEBUG[26209]: pbx.c:4786 __ast_pbx_run: Spawn extension
(internal,901148746612254,1) exited non-zero on 'SIP/11-00000002'
== Spawn extension (internal, 901148746612254, 1) exited non-zero on
'SIP/11-00000002'
[Sep 5 14:04:53] DEBUG[26209]: channel.c:2679 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/11-00000002'
[Sep 5 14:04:53] DEBUG[26209]: channel.c:2807 ast_hangup: Hanging up channel
'SIP/11-00000002'
[Sep 5 14:04:53] DEBUG[26209]: chan_sip.c:6096 sip_hangup: Hangup call
SIP/11-00000002, SIP callid [email protected]
[Sep 5 14:04:53] DEBUG[26209]: res_rtp_asterisk.c:2393
ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x885bf68'
[Sep 5 14:04:53] DEBUG[26083]: chan_sip.c:4012 __sip_ack: Stopping
retransmission on '[email protected]' of Request 102: Match Found
[Sep 5 14:04:53] DEBUG[26083]: rtp_engine.c:295 instance_destructor: Destroyed
RTP instance '0x885bf68'
[Sep 5 14:04:54] DEBUG[26085]: chan_iax2.c:2393 peercnt_remove: ip callno
count decremented to 1 for 8.14.120.23
[Sep 5 14:04:54] DEBUG[26094]: chan_iax2.c:2363 peercnt_add: ip callno count
incremented to 2 for 8.14.120.23
[Sep 5 14:04:54] DEBUG[26095]: chan_iax2.c:2711 sched_delay_remove: schedule
decrement of callno used for 8.14.120.23 in 60 seconds
--
Joseph
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
