Re: [asterisk-users] Asterisk 1.8 not accepting call from DID
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= 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/11-0002, 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-1270-0003' with that of 'SIP/11-0002' [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
[asterisk-users] Asterisk 1.8 not accepting call from DID
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= 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/11-0002, 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-1270-0003' with that of 'SIP/11-0002' [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-1270-0003 is making progress passing it to SIP/11-0002 [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1542 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/11-0002' with that of 'SIP/pstn-1270-0003' [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]:
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 from10.0.0.110:6000 (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= 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/11-0002, 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-1270-0003' with that of 'SIP/11-0002' [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-1270-0003 is making
