Hello Jurijs,
It worked. You saved my day. Can't thank you enough
Best Regards,
Jesse
At 2017-09-22 16:51:32, "Jurijs Ivolga" <[email protected]> wrote:
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 <[email protected]> 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" <[email protected]> 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 <[email protected]> wrote:
Hello,
No luck. Still the same. Here goes the full log, sorry if it's a little
overwhelming
------------------------------------------------------------------------
INVITE sip:[email protected]: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:[email protected]>;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
To: <sip:[email protected]>
Contact:
<sip:[email protected]: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:[email protected]>;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
To: <sip:[email protected]>
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/[email protected] [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/[email protected] 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/[email protected] [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:[email protected]>;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
To: <sip:[email protected]>;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:[email protected]>;party=calling;privacy=off;screen=no
------------------------------------------------------------------------
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1
(sofia/internal/[email protected]) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel
sofia/internal/[email protected] [CS_DESTROY]
recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597:
------------------------------------------------------------------------
ACK sip:[email protected]: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:[email protected]>;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
To: <sip:[email protected]>;tag=3N0c8m5X06NBj
Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe
CSeq: 21643 ACK
Content-Length: 0
------------------------------------------------------------------------
At 2017-09-22 16:14:37, "Jurijs Ivolga" <[email protected]> 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 <[email protected]> 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 <[email protected]> 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/[email protected] [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/[email protected]
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" <[email protected]> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number" expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <[email protected]> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided
Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <[email protected]> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <[email protected]> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert
Record-Route, try in following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <[email protected]> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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" <[email protected]> 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/[email protected]"/>
<action application="playback" data="ivr/ivr-user_busy.wav"/>
</condition>
</extension>
ср, 20 сент. 2017 г. в 10:14, 赵国杰 <[email protected]>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <[email protected]> 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 <[email protected]>:
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" <[email protected]> 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="[email protected]";
> route(RELAY);
> exit;
>}
>
>_______________________________________________
>Kamailio (SER) - Users Mailing List
>[email protected]
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