Hello Everyone, I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ? FLow : *SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider* NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side. Currently I can see below log in freeswitch logs : m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812@192.168.1.26 Action bridge(sofia/gateway/localgateway/001***) m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv I tried different solutions for this issue but no luck. 1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i) 2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i) 3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000 @@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000 @40i,PCMA@32000@40i 2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false 3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false 4.outbound-codec-prefs=PCMU at40i,PCMA at 40i 5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i ,inbound-late-negotiation=false 6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false Please point me where i am doing wrong. -- Thanks, Bhavik Patel
I know you can do this easyly with opensips 22 авг. 2015 г. 16:10 пользователь "bhavik patel" < bhavikpatel14388@gmail.com> написал:
Hello Everyone,
I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ?
FLow : *SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider*
NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side.
Currently I can see below log in freeswitch logs :
m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812@192.168.1.26 Action bridge(sofia/gateway/localgateway/001***)
m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
I tried different solutions for this issue but no luck.
1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i)
2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i)
3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000 @@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000 @40i,PCMA@32000@40i
2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false
3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false
4.outbound-codec-prefs=PCMU at40i,PCMA at 40i
5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i ,inbound-late-negotiation=false
6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false
Please point me where i am doing wrong.
-- Thanks, Bhavik Patel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Hi Bhahvik, We have instances were ptime:30 is signalled but the stream is actually ptime:20 so we override the signalling with this: absolute_codec_string=PCMA@20i,PCMU@20i? It is slightly different to your examples below but 3 and 4 are close - I've used @ instead of "at". Hope this helps, David ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Kamil Nigmatullin <kamil.nigmatullin@gmail.com> Sent: Saturday, 22 August 2015 22:57 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Change ptime value I know you can do this easyly with opensips 22 ???. 2015 ?. 16:10 ???????????? "bhavik patel" <bhavikpatel14388@gmail.com<mailto:bhavikpatel14388@gmail.com>> ???????: Hello Everyone, I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ? FLow : SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side. Currently I can see below log in freeswitch logs : m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812@192.168.1.26<mailto:4602534812@192.168.1.26> Action bridge(sofia/gateway/localgateway/001***) m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv I tried different solutions for this issue but no luck. 1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i) 2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i) 3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000@@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000@40i,PCMA@32000@40i 2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false 3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false 4.outbound-codec-prefs=PCMU at40i,PCMA at 40i 5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i,inbound-late-negotiation=false 6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false Please point me where i am doing wrong. -- Thanks, Bhavik Patel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Is it possible i can change it for G729 codec ? Like G729@40i ? On my Server G729 as Pass-through mode. On Mon, Aug 24, 2015 at 6:40 AM, David Witham <david.witham@netsip.com.au> wrote:
Hi Bhahvik,
We have instances were ptime:30 is signalled but the stream is actually ptime:20 so we override the signalling with this:
absolute_codec_string=PCMA@20i,PCMU@20i
It is slightly different to your examples below but 3 and 4 are close - I've used @ instead of "at".
Hope this helps,
David ------------------------------ *From:* freeswitch-users-bounces@lists.freeswitch.org < freeswitch-users-bounces@lists.freeswitch.org> on behalf of Kamil Nigmatullin <kamil.nigmatullin@gmail.com> *Sent:* Saturday, 22 August 2015 22:57 *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Change ptime value
I know you can do this easyly with opensips 22 авг. 2015 г. 16:10 пользователь "bhavik patel" < bhavikpatel14388@gmail.com> написал:
Hello Everyone,
I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ?
FLow : *SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider*
NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side.
Currently I can see below log in freeswitch logs :
m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812@192.168.1.26 Action bridge(sofia/gateway/localgateway/001***)
m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
I tried different solutions for this issue but no luck.
1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i)
2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i)
3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000 @@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000 @40i,PCMA@32000@40i
2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false
3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false
4.outbound-codec-prefs=PCMU at40i,PCMA at 40i
5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i ,inbound-late-negotiation=false
6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false
Please point me where i am doing wrong.
-- Thanks, Bhavik Patel
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Thanks, Bhavik Patel
If you're already passing G729 through, including G729@40i in the list should override the ptime. Check the SDP on the outbound leg of the call to confirm. ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@lists.freeswitch.org> on behalf of bhavik patel <bhavikpatel14388@gmail.com> Sent: Monday, 24 August 2015 17:29 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Change ptime value Is it possible i can change it for G729 codec ? Like G729@40i ? On my Server G729 as Pass-through mode. On Mon, Aug 24, 2015 at 6:40 AM, David Witham <david.witham@netsip.com.au<mailto:david.witham@netsip.com.au>> wrote: Hi Bhahvik, We have instances were ptime:30 is signalled but the stream is actually ptime:20 so we override the signalling with this: absolute_codec_string=PCMA@20i,PCMU@20i It is slightly different to your examples below but 3 and 4 are close - I've used @ instead of "at". Hope this helps, David ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> <freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> on behalf of Kamil Nigmatullin <kamil.nigmatullin@gmail.com<mailto:kamil.nigmatullin@gmail.com>> Sent: Saturday, 22 August 2015 22:57 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Change ptime value I know you can do this easyly with opensips 22 авг. 2015 г. 16:10 пользователь "bhavik patel" <bhavikpatel14388@gmail.com<mailto:bhavikpatel14388@gmail.com>> написал: Hello Everyone, I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ? FLow : SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side. Currently I can see below log in freeswitch logs : m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812@192.168.1.26<mailto:4602534812@192.168.1.26> Action bridge(sofia/gateway/localgateway/001***) m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv I tried different solutions for this issue but no luck. 1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i) 2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i) 3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000@@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000@40i,PCMA@32000@40i 2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false 3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false 4.outbound-codec-prefs=PCMU at40i,PCMA at 40i 5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i,inbound-late-negotiation=false 6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false Please point me where i am doing wrong. -- Thanks, Bhavik Patel _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Thanks, Bhavik Patel
Is anyone using smartos or solaris and using tcp for gateways? I have raised a ticket https://freeswitch.org/jira/browse/FS-7991 When trying to register to the gateway I get: 2015-08-19 15:07:59.194456 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #1 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:00.174456 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (79662368) nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all(0, 0, fffffd7ffe904d20) nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set 2015-08-19 15:08:31.174457 [NOTICE] sofia_reg.c:448 Registering 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 nua.c:622 nua_register() nua: nua_register: entering nua_stack.c:529 nua_signal() nua(1dc6310): sent signal r_register nua_stack.c:569 nua_stack_signal() nua(1dc6310): recv signal r_register nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::936c10, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(1dc6310): adding register usage nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(2cc6850) nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com.") returned 1 entries nta.c:10598 outgoing_query_srv() nta: for "manchester.out.simwood.com" query "_sip._tcp.manchester.out.simwood.com" SRV (cached) sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com.") returned 1 entries nta.c:10803 outgoing_query_a() nta: for "manchester.out.simwood.com" query "manchester.out.simwood.com" A (cached) nta.c:10856 outgoing_answer_a() nta: manchester.out.simwood.com. IN A 178.22.143.66 tport.c:3257 tport_tsend() tport_tsend(1555810) tpn = tcp/178.22.143.66:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 178.22.143.66:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(1555810): not found by name tcp/178.22.143.66:5060 tport.c:862 tport_alloc_secondary() tport_alloc_secondary(1555810): new secondary tport 151fa10 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPINTVL to 30 tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory tport.c:3373 tport_tsend() tport_socket failed in tsend nta.c:8496 outgoing_print_tport_error() nta: REGISTER (79662384): No such file or directory (2) with tcp/[178.22.143.66]:5060 nta.c:1350 set_timeout() nta: timer set to 32000 ms nta.c:1348 set_timeout() nta: timer shortened to 5000 ms nua_stack.c:271 nua_stack_event() nua(1dc6310): event r_register 503 Service Unavailable nua_dialog.c:397 nua_dialog_usage_remove_at() nua(1dc6310): removing register usage nta.c:4470 nta_leg_destroy() nta_leg_destroy(2cc6850) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:31.174457 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #2 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:32.174463 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. No SIP is ever sent. It seems to go bad here: tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory If I change the transport to UDP, sip flows. Anyone any ideas? TIA Daz
It looks like it attempts to open a tcp socket and fails. I'd check system limits on tcp sockets, and settings related to reusing sockets (you would have to go back in the log and confirm that socket was recently used) and see how those things might be different on SmartOS.
On Aug 22, 2015, at 10:53 AM, Support <support@directvoip.co.uk> wrote:
Is anyone using smartos or solaris and using tcp for gateways?
I have raised a ticket https://freeswitch.org/jira/browse/FS-7991 <https://freeswitch.org/jira/browse/FS-7991>
When trying to register to the gateway I get:
2015-08-19 15:07:59.194456 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #1 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:00.174456 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (79662368) nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all(0, 0, fffffd7ffe904d20) nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set 2015-08-19 15:08:31.174457 [NOTICE] sofia_reg.c:448 Registering 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 nua.c:622 nua_register() nua: nua_register: entering nua_stack.c:529 nua_signal() nua(1dc6310): sent signal r_register nua_stack.c:569 nua_stack_signal() nua(1dc6310): recv signal r_register nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::936c10, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(1dc6310): adding register usage nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(2cc6850) nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com <http://tcp.manchester.out.simwood.com/>.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com <http://tcp.manchester.out.simwood.com/>.") returned 1 entries nta.c:10598 outgoing_query_srv() nta: for "manchester.out.simwood.com <http://manchester.out.simwood.com/>" query "_sip._tcp.manchester.out.simwood.com <http://tcp.manchester.out.simwood.com/>" SRV (cached) sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com <http://manchester.out.simwood.com/>.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com <http://manchester.out.simwood.com/>.") returned 1 entries nta.c:10803 outgoing_query_a() nta: for "manchester.out.simwood.com <http://manchester.out.simwood.com/>" query "manchester.out.simwood.com <http://manchester.out.simwood.com/>" A (cached) nta.c:10856 outgoing_answer_a() nta: manchester.out.simwood.com <http://manchester.out.simwood.com/>. IN A 178.22.143.66 tport.c:3257 tport_tsend() tport_tsend(1555810) tpn = tcp/178.22.143.66:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 178.22.143.66:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(1555810): not found by name tcp/178.22.143.66:5060 tport.c:862 tport_alloc_secondary() tport_alloc_secondary(1555810): new secondary tport 151fa10 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPINTVL to 30 tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory tport.c:3373 tport_tsend() tport_socket failed in tsend nta.c:8496 outgoing_print_tport_error() nta: REGISTER (79662384): No such file or directory (2) with tcp/[178.22.143.66]:5060 nta.c:1350 set_timeout() nta: timer set to 32000 ms nta.c:1348 set_timeout() nta: timer shortened to 5000 ms nua_stack.c:271 nua_stack_event() nua(1dc6310): event r_register 503 Service Unavailable nua_dialog.c:397 nua_dialog_usage_remove_at() nua(1dc6310): removing register usage nta.c:4470 nta_leg_destroy() nta_leg_destroy(2cc6850) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:31.174457 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #2 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:32.174463 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. No SIP is ever sent. It seems to go bad here: tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory If I change the transport to UDP, sip flows. Anyone any ideas? TIA Daz
Michael, Thanks for the reply. I'd just like to add that from startup with a fresh install of freeswitch this is happening. Another strange thing is that connecting handsets using TCP to the internal profile seems to present no problems at all. Regards Darren _____ From: Michael Jerris [mailto:mike@jerris.com] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Sat, 22 Aug 2015 15:09:40 +0000 Subject: Re: [Freeswitch-users] SmartOS or Solaris - using tcp for gateways It looks like it attempts to open a tcp socket and fails. I'd check system limits on tcp sockets, and settings related to reusing sockets (you would have to go back in the log and confirm that socket was recently used) and see how those things might be different on SmartOS. On Aug 22, 2015, at 10:53 AM, Support <support@directvoip.co.uk> wrote: Is anyone using smartos or solaris and using tcp for gateways? I have raised a ticket https://freeswitch.org/jira/browse/FS-7991 When trying to register to the gateway I get: 2015-08-19 15:07:59.194456 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #1 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:00.174456 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER (79662368) nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all(0, 0, fffffd7ffe904d20) nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set 2015-08-19 15:08:31.174457 [NOTICE] sofia_reg.c:448 Registering 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 nua.c:622 nua_register() nua: nua_register: entering nua_stack.c:529 nua_signal() nua(1dc6310): sent signal r_register nua_stack.c:569 nua_stack_signal() nua(1dc6310): recv signal r_register nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::936c10, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(1dc6310): adding register usage nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(2cc6850) nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, SRV, "_sip._tcp.manchester.out.simwood.com.") returned 1 entries nta.c:10598 outgoing_query_srv() nta: for "manchester.out.simwood.com" query "_sip._tcp.manchester.out.simwood.com" SRV (cached) sres_cache.c:272 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com.") called sres_cache.c:318 sres_cache_get() sres_cache_get(149e7a0, A, "manchester.out.simwood.com.") returned 1 entries nta.c:10803 outgoing_query_a() nta: for "manchester.out.simwood.com" query "manchester.out.simwood.com" A (cached) nta.c:10856 outgoing_answer_a() nta: manchester.out.simwood.com. IN A 178.22.143.66 tport.c:3257 tport_tsend() tport_tsend(1555810) tpn = tcp/178.22.143.66:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 178.22.143.66:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(1555810): not found by name tcp/178.22.143.66:5060 tport.c:862 tport_alloc_secondary() tport_alloc_secondary(1555810): new secondary tport 151fa10 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(151fa10): Setting TCP_KEEPINTVL to 30 tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory tport.c:3373 tport_tsend() tport_socket failed in tsend nta.c:8496 outgoing_print_tport_error() nta: REGISTER (79662384): No such file or directory (2) with tcp/[178.22.143.66]:5060 nta.c:1350 set_timeout() nta: timer set to 32000 ms nta.c:1348 set_timeout() nta: timer shortened to 5000 ms nua_stack.c:271 nua_stack_event() nua(1dc6310): event r_register 503 Service Unavailable nua_dialog.c:397 nua_dialog_usage_remove_at() nua(1dc6310): removing register usage nta.c:4470 nta_leg_destroy() nta_leg_destroy(2cc6850) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:31.174457 [ERR] sofia_reg.c:2367 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration with status Service Unavailable [503]. failure #2 nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2015-08-19 15:08:32.174463 [WARNING] sofia_reg.c:505 97b1323c-9ff1-44e3-b3b2-0e92fefa62a7 Failed Registration [503], setting retry to 30 seconds. No SIP is ever sent. It seems to go bad here: tport.c:1017 tport_base_connect() tport_base_connect(1555810): connect(pf=2 tcp/[178.22.143.66]:5060): No such file or directory If I change the transport to UDP, sip flows. Anyone any ideas? TIA Daz
participants (5)
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bhavik patel
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David Witham
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Kamil Nigmatullin
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Michael Jerris
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Support