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=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alias=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:BrFCcbuKqPea6vy8L9Imh6dqhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpamPBj6prelcsjywL+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=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;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=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB 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+git~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-ac89eb6ccf78] 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=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB 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+git~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=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB 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.ivolga@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.ivolga@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 zhaoguojie2010@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.ivolga@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($rU=="12345") { if(is_method("INVITE")) { record_route(); $ru = "sip:" + "offline" + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; } }
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello I added below code to let kamailio route invite to freeswitch: if ($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/sounds/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.safarov@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/3000@example.org"/> <action application="playback" data="ivr/ivr-user_busy.wav"/> </condition> </extension>
ср, 20 сент. 2017 г. в 10:14, 赵国杰 zhaoguojie2010@163.com:
Hello Sergey, I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" s.safarov@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 zhaoguojie2010@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.tryba@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="busymessage@asterisk.example.org"; route(RELAY); exit; }
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