Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

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  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"  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  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: 
>>Record-Route: 
>>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: ;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>>To: 
>>Contact: > 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: 
>>Record-Route: 
>>From: ;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>>To: 
>>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@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@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 

Re: [SR-Users] how to play ring tune when callee declines

[赵国杰]

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"  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  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: 
   Record-Route: 
   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: ;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
   To: 
   Contact: 

   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: 
   Record-Route: 
   From: ;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
   To: 
   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 

Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

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 
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  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: 
>>Record-Route: 
>>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: ;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>>To: 
>>Contact: > 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: 
>>Record-Route: 
>>From: ;tag=MW06SkJdOZIiqvT4T9DFn0X5
>> QazXW6BB
>>To: 
>>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@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@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]
>> 

Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

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  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: 
>Record-Route: 
>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: ;tag=
> MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
>To: 
>Contact:  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: 
>Record-Route: 
>From: ;tag=
> MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
>To: 
>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=
> 

Re: [SR-Users] how to play ring tune when callee declines

[赵国杰]

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: 
   Record-Route: 
   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: ;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
   To: 
   Contact: 

   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: 
   Record-Route: 
   From: ;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
   To: 
   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: ;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB
   To: ;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, 

Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

Hi,

You probably don't need record route and you need to remove ""

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

  

  


Jurijs

On Fri, Sep 22, 2017 at 10:54 AM, 赵国杰  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
> 
>   
> 
> 
>   
> 
>
> *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:
> ,
> 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"  wrote:
>
> Hi,
>
> You need to add:
>
>  
>   
> 
>   
> 
>
> 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 
> 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  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"  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)
>>> + ":" +
>>> $sel(cfg_get.voicemail.srv_port);
>>> route(RELAY);
>>> exit;
>>> }
>>> }
>>>
>>> With kind regards,
>>>
>>> Jurijs
>>>
>>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰  wrote:
>>>
 Hello
 I added below code to let kamailio route invite to freeswitch:
 if ($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
  
   
 
 
   
 

 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 

Re: [SR-Users] how to play ring tune when callee declines

[赵国杰]

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

  
 

  



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: 
, 
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"  wrote:

Hi,


You need to add:

 
  

  



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  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  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"  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  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
 
  


  



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"  wrote:

You can add this example to dialplan and make test



  



  





ср, 20 сент. 2017 г. в 10:14, 赵国杰 :

Hello Sergey,
 I installed freeswitch, what should i do next?







At 2017-09-19 12:07:23, "Sergey Safarov"  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 :

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" 

Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

Hi,

You need to add:

 
  

  


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 
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  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"  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)
>> + ":" +
>> $sel(cfg_get.voicemail.srv_port);
>> route(RELAY);
>> exit;
>> }
>> }
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰  wrote:
>>
>>> Hello
>>> I added below code to let kamailio route invite to freeswitch:
>>> if ($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
>>>  
>>>   
>>> 
>>> 
>>>   
>>> 
>>>
>>> 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"  wrote:
>>>
>>> You can add this example to dialplan and make test
>>>
>>> 
>>>   
>>> 
>>> 
>>> 
>>>   
>>> 
>>>
>>>
>>> ср, 20 сент. 2017 г. в 10:14, 赵国杰 :
>>>
 Hello Sergey,
  I installed freeswitch, what should i do next?





 At 2017-09-19 12:07:23, "Sergey Safarov"  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 :

> 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"  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

>>>
>>>

Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

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  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"  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)
> + ":" +
> $sel(cfg_get.voicemail.srv_port);
> route(RELAY);
> exit;
> }
> }
>
> With kind regards,
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰  wrote:
>
>> Hello
>> I added below code to let kamailio route invite to freeswitch:
>> if ($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
>>  
>>   
>> 
>> 
>>   
>> 
>>
>> 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"  wrote:
>>
>> You can add this example to dialplan and make test
>>
>> 
>>   
>> 
>> 
>> 
>>   
>> 
>>
>>
>> ср, 20 сент. 2017 г. в 10:14, 赵国杰 :
>>
>>> Hello Sergey,
>>>  I installed freeswitch, what should i do next?
>>>
>>>
>>>
>>>
>>>
>>> At 2017-09-19 12:07:23, "Sergey Safarov"  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 :
>>>
 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"  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;
 >}
 >
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 >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users




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>>>
>>>
>>>
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Re: [SR-Users] how to play ring tune when callee declines

[赵国杰]

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"  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  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
 
  


  



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"  wrote:

You can add this example to dialplan and make test



  



  





ср, 20 сент. 2017 г. в 10:14, 赵国杰 :

Hello Sergey,
 I installed freeswitch, what should i do next?







At 2017-09-19 12:07:23, "Sergey Safarov"  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 :

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"  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




 

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Re: [SR-Users] how to play ring tune when callee declines

[Jurijs Ivolga]

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  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
>  
>   
> 
> 
>   
> 
>
> 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"  wrote:
>
> You can add this example to dialplan and make test
>
> 
>   
> 
> 
> 
>   
> 
>
>
> ср, 20 сент. 2017 г. в 10:14, 赵国杰 :
>
>> Hello Sergey,
>>  I installed freeswitch, what should i do next?
>>
>>
>>
>>
>>
>> At 2017-09-19 12:07:23, "Sergey Safarov"  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 :
>>
>>> 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"  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
>>>
>>>
>>>
>>>
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>>> sr-users@lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>>
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
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>
>
>
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Re: [SR-Users] how to play ring tune when callee declines

[赵国杰]

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
 
  


  



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"  wrote:

You can add this example to dialplan and make test



  



  





ср, 20 сент. 2017 г. в 10:14, 赵国杰 :

Hello Sergey,
 I installed freeswitch, what should i do next?







At 2017-09-19 12:07:23, "Sergey Safarov"  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 :

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"  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




 

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Re: [SR-Users] how to play ring tune when callee declines

[Sergey Safarov]

You can add this example to dialplan and make test


  



  



ср, 20 сент. 2017 г. в 10:14, 赵国杰 :

> Hello Sergey,
>  I installed freeswitch, what should i do next?
>
>
>
>
>
> At 2017-09-19 12:07:23, "Sergey Safarov"  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 :
>
>> 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"  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
>>
>>
>>
>>
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>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
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Re: [SR-Users] how to play ring tune when callee declines

[Daniel Tryba]

On Tue, Sep 19, 2017 at 11:26:46AM +0800, 赵国杰 wrote:
> 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.

Just use asterisk/freeswitch/sems/whatever to playback media and you don't
have to touch SDP in kamailio to do this. In order to avoid answering
the call you have to generate the audio without a 200 OK, so your option
is 183 Progress, which asterisk will do for you with Playback(foo,noanswer).
 

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Re: [SR-Users] how to play ring tune when callee declines

[Sergey Safarov]

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 :

> 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"  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
>
>
>
>
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[SR-Users] how to play ring tune when callee declines

[赵国杰]

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"  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;
>}
>
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