[asterisk-users] Asterisk 1.8 not accepting call from DID
Joseph
syscon780 at gmail.com
Mon Sep 5 20:50:34 CDT 2011
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 at internal:1] Dial("SIP/11-00000002", "SIP/901148746612254 at 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 5a2cdf8339e0ad2911ad393036c05165 at 127.0.0.1:0 - 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
>770d283f78ef7d00782d2dd043212ed2 at 10.0.0.103:5060
> -- Called SIP/901148746612254 at pstn-1270
>[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on
>'770d283f78ef7d00782d2dd043212ed2 at 10.0.0.103:5060' Request 102: Found
>[Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on
>'770d283f78ef7d00782d2dd043212ed2 at 10.0.0.103:5060' 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 '770d283f78ef7d00782d2dd043212ed2 at 10.0.0.103:5060' 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 '9320679215920111346 at 10.0.0.110' 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 at 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 770d283f78ef7d00782d2dd043212ed2 at 10.0.0.103:5060
>[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 9320679215920111346 at 10.0.0.110
>[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 '9320679215920111346 at 10.0.0.110' 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
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