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=z9hG4bK04d4.
> 2c5c86a459371d838623651e8f5b6984.0;i=1
> Via: SIP/2.0/TLS 10.60.208.121:43603;received=
> 175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U
> 0w0JRcTLD9Y;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=z9hG4bKPj4dLalct0388uwB380xv2U
> 0w0JRcTLD9Y;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=z9hG4bKPj4dLalct0388uwB380xv2U
> 0w0JRcTLD9Y;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.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
>
>
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