Hi, First try to set variable in vars.xml, as I sent if didn't help, you can try to turn encryption off on your CSipSimple
With kind regards, Jurijs On Fri, Sep 22, 2017 at 11:43 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: > > Thanks man, > I didn't explicitly set srtp in kamailio nor freeswitch, how do i turn > it off? > > > At 2017-09-22 16:32:10, "Jurijs Ivolga" <jurijs.ivo...@gmail.com> wrote: > > Hi, > > 1) You need to change default password!!!!!!!!!!!! > *"Open /usr/local/freeswitch/conf/**vars.xml and change the > default_password."* > > 2) You are calling into Freeswitch with encryption on and probably of this > your call is failing, maybe you can try first to try without SRTP and if it > works, then you can try to make it work with SRTP > > With kind regards, > > Jurijs > > On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: > >> >> Hello, >> No luck. Still the same. Here goes the full log, sorry if it's a >> little overwhelming >> >> ------------------------------------------------------------------------ >> INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 >> Record-Route: <sip:35.202.167.70:5060;r2=on;lr;nat=yes> >> Record-Route: <sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes> >> Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG >> 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 >> Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 >> 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 >> xv2U0w0JRcTLD9Y;alias >> Max-Forwards: 69 >> From: <sip:13112345678@35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5 >> QazXW6BB >> To: <sip:12345@35.202.167.70> >> Contact: <sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alia >> s=175.100.202.254~33189~3> >> Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe >> CSeq: 21643 INVITE >> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, >> NOTIFY, REFER, MESSAGE, OPTIONS >> Supported: replaces, 100rel, timer, norefersub >> Session-Expires: 1800 >> Min-SE: 90 >> User-Agent: CSipSimple_HWNXT-24/r2457 >> Content-Type: application/sdp >> Content-Length: 515 >> >> v=0 >> o=- 3715057398 3715057398 IN IP4 35.185.130.154 >> s=pjmedia >> c=IN IP4 35.185.130.154 >> t=0 0 >> m=audio 40026 RTP/AVP 9 8 0 106 101 >> c=IN IP4 35.185.130.154 >> a=rtcp:40027 >> a=sendrecv >> a=rtpmap:9 G722/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:106 speex/16000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6d >> qhorYovx1RdXKlLsP >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpa >> mPBj6prelcsjywL+M >> a=nortpproxy:yes >> ----------------------------------------------------------- >> ------------- >> send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105: >> ----------------------------------------------------------- >> ------------- >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG >> 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 >> Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 >> 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 >> xv2U0w0JRcTLD9Y;alias >> Record-Route: <sip:35.202.167.70:5060;r2=on;lr;nat=yes> >> Record-Route: <sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes> >> From: <sip:13112345678@35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5 >> QazXW6BB >> To: <sip:12345@35.202.167.70> >> Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe >> CSeq: 21643 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi >> t~20160205T175853Z~ca9207aa32~64bit >> Content-Length: 0 >> >> ----------------------------------------------------------- >> ------------- >> 2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel >> sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89e >> b6ccf78] >> 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing >> 13112345678 <13112345678>->prompt-1000 in context public >> 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer >> sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] >> 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing >> 13112345678 <13112345678>->prompt-1000 in context default >> 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING >> WARNING WARNING WARNING WARNING WARNING WARNING WARNING >> 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open >> /usr/local/freeswitch/conf/vars.xml and change the default_password. >> 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type >> 'reloadxml' at the console. >> 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING >> WARNING WARNING WARNING WARNING WARNING WARNING WARNING >> 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in >> RTP/AVP, refer to rfc3711 >> 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup >> sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] >> [INCOMPATIBLE_DESTINATION] >> send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628: >> ----------------------------------------------------------- >> ------------- >> SIP/2.0 488 Not Acceptable Here >> Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG >> 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 >> Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 >> 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 >> xv2U0w0JRcTLD9Y;alias >> Max-Forwards: 68 >> From: <sip:13112345678@35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5 >> QazXW6BB >> To: <sip:12345@35.202.167.70>;tag=3N0c8m5X06NBj >> Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe >> CSeq: 21643 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi >> t~20160205T175853Z~ca9207aa32~64bit >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, presence, as-feature-event, >> dialog, line-seize, call-info, sla, include-session-description, >> presence.winfo, message-summary, refer >> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> Content-Length: 0 >> Remote-Party-ID: "prompt-1000" <sip:prompt-1000@35.202.167.70 >> >;party=calling;privacy=off;screen=no >> >> ----------------------------------------------------------- >> ------------- >> 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 >> (sofia/internal/13112345678@35.202.167.70) Ended >> 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close >> Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] >> recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597: >> ----------------------------------------------------------- >> ------------- >> ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 >> Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG >> 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 >> Max-Forwards: 69 >> From: <sip:13112345678@35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5 >> QazXW6BB >> To: <sip:12345@35.202.167.70>;tag=3N0c8m5X06NBj >> Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe >> CSeq: 21643 ACK >> Content-Length: 0 >> >> ----------------------------------------------------------- >> ------------- >> >> >> At 2017-09-22 16:14:37, "Jurijs Ivolga" <jurijs.ivo...@gmail.com> wrote: >> >> Hi, >> >> You need to answer call too... >> >> Try this: >> >> * in freeswitch/conf/dialplan/default.xml* >> <extension name="prompt-offline"> >> <condition field="destination_number" expression="^prompt-(.+)$"> >> >> <action application="answer"/> >> >> <action application="playback" data="ivr/ivr-user_busy.wav"/> >> </condition> >> </extension> >> >> Please send full logs next time, you can remove IP-addresses and other info, >> but one line is not really helpful. >> >> With kind regards, >> >> >> Jurijs >> >> On Fri, Sep 22, 2017 at 11:00 AM, Jurijs Ivolga <jurijs.ivo...@gmail.com> >> wrote: >> >>> Hi, >>> >>> You probably don't need record route and you need to remove "<action >>> application="bridge" data="user/$1@${domain_name}"/>" >>> >>> Try in this way: >>> >>> *In kamailio.cfg* I added if ($rU=="12345") { >>> if(is_method("INVITE")) { >>> #record_route(); >>> $ru = "sip:prompt-1000@" + >>> $sel(cfg_get.voicemail.srv_ip) >>> + ":" + >>> $sel(cfg_get.voicemail.srv_port); >>> route(RELAY); >>> exit; >>> } >>> } >>> >>> * in freeswitch/conf/dialplan/default.xml*, i added >>> <extension name="prompt-offline"> >>> <condition field="destination_number" expression="^prompt-(.+)$"> >>> <action application="playback" data="ivr/ivr-user_busy.wav"/> >>> </condition> >>> </extension> >>> >>> Jurijs >>> >>> On Fri, Sep 22, 2017 at 10:54 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: >>> >>>> Hi guy. >>>> sorry for the confusion. I'll try to reorganize it. >>>> >>>> * In kamailio.cfg* I added >>>> if ($rU=="12345") { >>>> if(is_method("INVITE")) { >>>> #record_route(); >>>> $ru = "sip:prompt-1000@" + >>>> $sel(cfg_get.voicemail.srv_ip) >>>> + ":" + >>>> $sel(cfg_get.voicemail.srv_port); >>>> route(RELAY); >>>> exit; >>>> } >>>> } >>>> >>>> * in freeswitch/conf/dialplan/default.xml*, i added >>>> <extension name="prompt-offline"> >>>> <condition field="destination_number" expression="^prompt-(.+)$"> >>>> <action application="bridge" data="user/$1@${domain_name}"/> >>>> <action application="playback" data="ivr/ivr-user_busy.wav"/> >>>> </condition> >>>> </extension> >>>> >>>> *sofia log:* >>>> [NOTICE] switch_channel.c:1077 New Channel sofia/internal/ >>>> 13112345678@35.202.167.70 [848d0dd5-0513-41bd-982c-45a2a886e194] >>>> [INFO] mod_dialplan_xml.c:635 Processing 13112345678 >>>> <13112345678>->prompt-1000 in context public >>>> [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35. >>>> 202.167.70 to XML[prompt-1000@default] >>>> [INFO] mod_dialplan_xml.c:635 Processing 13112345678 >>>> <13112345678>->prompt-1000 in context default >>>> [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel >>>> of type [error] cause: [USER_NOT_REGISTERED] >>>> [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel >>>> of type [user] cause: [USER_NOT_REGISTERED] >>>> ----------------------------------------------------------- >>>> ------------- >>>> SIP/2.0 480 Temporarily Unavailable >>>> ...... >>>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED" >>>> >>>> ----------------------------------------------------------- >>>> ------------- >>>> >>>> However, if i delete: >>>> <action application="bridge" data="user/$1@${domain_name}"/>, >>>> the FS returns 488 instead of 480. Reason: >>>> Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>> >>>> Thanks >>>> >>>> >>>> >>>> >>>> At 2017-09-22 15:31:51, "Jurijs Ivolga" <jurijs.ivo...@gmail.com> >>>> wrote: >>>> >>>> Hi, >>>> >>>> You need to add: >>>> >>>> <extension name="prompt-offline"> >>>> <condition field="destination_number" expression="^offline$"> >>>> <action application="playback" data="/usr/local/freeswitch/so >>>> unds/music/8000/suite-espanola-op-47-leyenda.wav"/> >>>> </condition> >>>> </extension> >>>> >>>> to conf/dialplan/default.xml >>>> >>>> in your code, you had extra line what was sending a call to 1000 >>>> extension. >>>> >>>> With kind regards, >>>> >>>> Jurijs >>>> >>>> On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga < >>>> jurijs.ivo...@gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> So, problem is not related to record route but to config of freeswitch. >>>>> >>>>> Not sure what you wrote in mail above, but you need to add code what >>>>> provided Sergey to: >>>>> >>>>> /usr/local/freeswitch/conf/dialplan/default.xml >>>>> >>>>> With kind regards, >>>>> >>>>> Jurijs >>>>> >>>>> On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: >>>>> >>>>>> Hello, >>>>>> Thanks for the heads up. The siptrace does help. >>>>>> Now the FS returns(with or without record_route();): >>>>>> SIP/2.0 480 Temporarily Unavailable >>>>>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED" >>>>>> >>>>>> I have generate offline.xml under conf/directory/default. Where >>>>>> did i miss? >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivo...@gmail.com> >>>>>> wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> Sip trace from Freeswitch will help, but I think you need to insert >>>>>> Record-Route, try in following way: >>>>>> >>>>>> if ($rU=="12345") { >>>>>> if(is_method("INVITE")) { >>>>>> record_route(); >>>>>> $ru = "sip:" + "offline" + "@" + >>>>>> $sel(cfg_get.voicemail.srv_ip) >>>>>> + ":" + >>>>>> $sel(cfg_get.voicemail.srv_port); >>>>>> route(RELAY); >>>>>> exit; >>>>>> } >>>>>> } >>>>>> >>>>>> With kind regards, >>>>>> >>>>>> Jurijs >>>>>> >>>>>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: >>>>>> >>>>>>> Hello >>>>>>> I added below code to let kamailio route invite to freeswitch: >>>>>>> if ($rU=="12345") { >>>>>>> if(is_method("INVITE")) { >>>>>>> $ru = "sip:" + "offline" + "@" + >>>>>>> $sel(cfg_get.voicemail.srv_ip) >>>>>>> + ":" + >>>>>>> $sel(cfg_get.voicemail.srv_port); >>>>>>> route(RELAY); >>>>>>> exit; >>>>>>> } >>>>>>> } >>>>>>> >>>>>>> in freeswitch dialplan/default.xml, i added >>>>>>> <extension name="prompt-offline"> >>>>>>> <condition field="destination_number" expression="^offline$"> >>>>>>> <action application="bridge" data="user/1000@${domain_name} >>>>>>> "/> >>>>>>> <action application="playback" data="/usr/local/freeswitch/so >>>>>>> unds/music/8000/suite-espanola-op-47-leyenda.wav"/> >>>>>>> </condition> >>>>>>> </extension> >>>>>>> >>>>>>> when i dialed 12345 on sip client, I can see the invite package to >>>>>>> freeswitch, and that's it. No package coming back from freeswitch. >>>>>>> Eventually, the sip client timeout. I >>>>>>> was hoping that when i dial 12345, "suite-espanola-op-47-leyenda.wav" >>>>>>> will be played. What did i do wrong? >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> At 2017-09-20 19:32:14, "Sergey Safarov" <s.safa...@gmail.com> >>>>>>> wrote: >>>>>>> >>>>>>> You can add this example to dialplan and make test >>>>>>> >>>>>>> <extension name="call_user"> >>>>>>> <condition> >>>>>>> <action application="set" data="continue_on_fail=NORMAL_ >>>>>>> TEMPORARY_FAILURE,USER_BUSY,NO_ROUTE_DESTINATION,SUBSCRIBER_ >>>>>>> ABSENT"/> >>>>>>> <action application="bridge" data="user/3...@example.org"/> >>>>>>> <action application="playback" data="ivr/ivr-user_busy.wav"/> >>>>>>> </condition> >>>>>>> </extension> >>>>>>> >>>>>>> >>>>>>> ср, 20 сент. 2017 г. в 10:14, 赵国杰 <zhaoguojie2...@163.com>: >>>>>>> >>>>>>>> Hello Sergey, >>>>>>>> I installed freeswitch, what should i do next? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> At 2017-09-19 12:07:23, "Sergey Safarov" <s.safa...@gmail.com> >>>>>>>> wrote: >>>>>>>> >>>>>>>> This can be implemenred using freeswitch. >>>>>>>> Ping me directly after you install freeswith on linux and configure >>>>>>>> ssh remote access >>>>>>>> >>>>>>>> вт, 19 сент. 2017 г., 6:27 赵国杰 <zhaoguojie2...@163.com>: >>>>>>>> >>>>>>>>> Thanks Daniel, >>>>>>>>> I've done some digging, and from Andrew Prokop's blog, it says >>>>>>>>> this envolves early midia. Usually this is done by reply a 183 to the >>>>>>>>> caller with media ip and port in the SDP. This makes sense but i >>>>>>>>> still have >>>>>>>>> no idea how to generate 183 response with embedded SDP. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> At 2017-09-18 18:05:46, "Daniel Tryba" <d.tr...@pocos.nl> wrote: >>>>>>>>> >On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote: >>>>>>>>> >> I want the caller to play a short audio(like "the number your >>>>>>>>> >> are calling is busy") when the callee declines the call. How can i >>>>>>>>> >> do that? >>>>>>>>> > >>>>>>>>> >You need to check for the status codes in a failure route and then >>>>>>>>> >somehow generate audio somewhere, which is out of the scope of >>>>>>>>> >kamailio >>>>>>>>> >(maybe rtpproxy can do this, otherwise use something like asterisk): >>>>>>>>> > >>>>>>>>> >failure_route[MANAGE_FAILURE] { >>>>>>>>> >if (t_check_status("486")) >>>>>>>>> >{ >>>>>>>>> > $du=null; >>>>>>>>> > $ru="busymess...@asterisk.example.org"; >>>>>>>>> > route(RELAY); >>>>>>>>> > exit; >>>>>>>>> >} >>>>>>>>> > >>>>>>>>> >_______________________________________________ >>>>>>>>> >Kamailio (SER) - Users Mailing List >>>>>>>>> >sr-users@lists.kamailio.org >>>>>>>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users@lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>> sr-users@lists.kamailio.org >>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Kamailio (SER) - Users Mailing List >>>>>>> sr-users@lists.kamailio.org >>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users@lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>> >>>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users@lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >> >> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users@lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
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