Jurijs

On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 <zhaoguojie2010@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=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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