you didn't provide your "dialplan" for the incoming call context from_poland?
nor registration string?
could be a dial plan problem .. or codec issue.. as long as you register
"properly" the server has no problem with NAT.. it's a routing or codec issue i
think.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
----------------------------------------
> Date: Mon, 5 Sep 2011 19:50:34 -0600
> From: syscon...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8 not accepting call from DID
>
> 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 5a2cdf8339e0ad2911ad393036c05165@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@10.0.0.103:5060
> > -- 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
> >'770d283f78ef7d00782d2dd043212ed2@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@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@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@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@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@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@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@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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--
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