[SR-Users] how to play ring tune when callee declines
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Sep 22 10:41:38 CEST 2017
Hi,
Please check this:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2013-November/006889.html
Probably you need to set
rtp_allow_crypto_in_avp=true in vars.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:32 AM, Jurijs Ivolga <jurijs.ivolga at 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 <zhaoguojie2010 at 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 at 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 at 35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>> To: <sip:12345 at 35.202.167.70>
>> Contact: <sip:13112345678 at 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 at 35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>> To: <sip:12345 at 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 at 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 at 35.202.167.70 to XML[prompt-1000 at 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 at 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 at 35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>> To: <sip:12345 at 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 at 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 at 35.202.167.70) Ended
>> 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close
>> Channel sofia/internal/13112345678 at 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 at 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 at 35.202.167.70>;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>> To: <sip:12345 at 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 at 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 at 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 at 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 at 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 at 35.
>>>> 202.167.70 to XML[prompt-1000 at 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 at gmail.com>
>>>> wrote:
>>>>
>>>> Hi,
>>>>
>>>> You need to add:
>>>>
>>>> <extension name="prompt-offline">
>>>> <condition field="destination_number" expression="^offline$">
>>>> <action application="playback" data="/usr/local/freeswitch/so
>>>> unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
>>>> </condition>
>>>> </extension>
>>>>
>>>> to conf/dialplan/default.xml
>>>>
>>>> in your code, you had extra line what was sending a call to 1000
>>>> extension.
>>>>
>>>> With kind regards,
>>>>
>>>> Jurijs
>>>>
>>>> On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <
>>>> jurijs.ivolga at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> So, problem is not related to record route but to config of freeswitch.
>>>>>
>>>>> Not sure what you wrote in mail above, but you need to add code what
>>>>> provided Sergey to:
>>>>>
>>>>> /usr/local/freeswitch/conf/dialplan/default.xml
>>>>>
>>>>> With kind regards,
>>>>>
>>>>> Jurijs
>>>>>
>>>>> On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>> Thanks for the heads up. The siptrace does help.
>>>>>> Now the FS returns(with or without record_route();):
>>>>>> SIP/2.0 480 Temporarily Unavailable
>>>>>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>>>>>>
>>>>>> I have generate offline.xml under conf/directory/default. Where
>>>>>> did i miss?
>>>>>>
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Sip trace from Freeswitch will help, but I think you need to insert
>>>>>> Record-Route, try in following way:
>>>>>>
>>>>>> if ($rU=="12345") {
>>>>>> if(is_method("INVITE")) {
>>>>>> record_route();
>>>>>> $ru = "sip:" + "offline" + "@" +
>>>>>> $sel(cfg_get.voicemail.srv_ip)
>>>>>> + ":" +
>>>>>> $sel(cfg_get.voicemail.srv_port);
>>>>>> route(RELAY);
>>>>>> exit;
>>>>>> }
>>>>>> }
>>>>>>
>>>>>> With kind regards,
>>>>>>
>>>>>> Jurijs
>>>>>>
>>>>>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:
>>>>>>
>>>>>>> Hello
>>>>>>> I added below code to let kamailio route invite to freeswitch:
>>>>>>> if ($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.safarov at 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 at 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 at 163.com>:
>>>>>>>
>>>>>>>> Hello Sergey,
>>>>>>>> I installed freeswitch, what should i do next?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov at 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 at 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 at 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 at asterisk.example.org";
>>>>>>>>> > route(RELAY);
>>>>>>>>> > exit;
>>>>>>>>> >}
>>>>>>>>> >
>>>>>>>>> >_______________________________________________
>>>>>>>>> >Kamailio (SER) - Users Mailing List
>>>>>>>>> >sr-users at lists.kamailio.org
>>>>>>>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>>>> sr-users at lists.kamailio.org
>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>>> sr-users at lists.kamailio.org
>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>> sr-users at lists.kamailio.org
>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Kamailio (SER) - Users Mailing List
>>>>>> sr-users at lists.kamailio.org
>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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