Thanks Sammy, I will use pastebin.com next as you recommended.

Thanks

Abdul


________________________________
From: sr-users <[email protected]> on behalf of SamyGo 
<[email protected]>
Sent: Thursday, February 11, 2016 11:50 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC


Use pastebin.com<http://pastebin.com> or something ?

[http://pastebin.com/i/facebook.png]<http://pastebin.com/>

Pastebin.com - #1 paste tool since 2002!<http://pastebin.com/>
pastebin.com
Pastebin.com is the number one paste tool since 2002. Pastebin is a website 
where you can store text online for a set period of time.

On Feb 11, 2016 18:32, "malik sherif" 
<[email protected]<mailto:[email protected]>> wrote:

While the full debug log is being approved, I just copy and paste some of the 
log.


2016-02-11 11:38:42.469315 [DEBUG] switch_core_codec.c:246 
sofia/internal/[email protected]<mailto:[email protected]>
 Restore previous codec PCMU:0.
2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 
[email protected]<mailto:[email protected]>
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing 
for [email protected]<mailto:[email protected]> in inbox
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1946 Update MWI: Messages 
Waiting yes
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1947 Update MWI: Update 
Reason NEW
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1948 Update MWI: Message 
Account [email protected]<mailto:[email protected]>
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1949 Update MWI: Voice 
Message 12/0
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:2901 
sofia/internal/[email protected]<mailto:[email protected]>
 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:535 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State EXECUTE going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Running State Change CS_HANGUP
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Callstate Change ACTIVE -> HANGUP
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State HANGUP
2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 
sofia/internal/[email protected]<mailto:[email protected]>
 Overriding SIP cause 480 with 904 from the other leg
2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:413 Channel 
sofia/internal/[email protected]<mailto:[email protected]>
 hanging up, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 
sofia/internal/[email protected]<mailto:[email protected]>
 Standard HANGUP, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State HANGUP going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State Change CS_HANGUP -> CS_REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal 
sofia/internal/[email protected]<mailto:[email protected]>
 [BREAK]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Running State Change CS_REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 
sofia/internal/[email protected]<mailto:[email protected]>
 Standard REPORTING, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State REPORTING going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State Change CS_REPORTING -> CS_DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal 
sofia/internal/[email protected]<mailto:[email protected]>
 [BREAK]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Locked, Waiting on external entities
2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Ended
2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel 
sofia/internal/[email protected]<mailto:[email protected]>
 [CS_DESTROY]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 
(sofia/internal/[email protected]<mailto:[email protected]>)
 Running State Change CS_DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State DESTROY
2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 
sofia/internal/[email protected]<mailto:[email protected]>
 SOFIA DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 
sofia/internal/[email protected]<mailto:[email protected]>
 Standard DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 
(sofia/internal/[email protected]<mailto:[email protected]>)
 State DESTROY going to sleep



________________________________
From: sr-users 
<[email protected]<mailto:[email protected]>>
 on behalf of SamyGo <[email protected]<mailto:[email protected]>>
Sent: Thursday, February 11, 2016 5:41 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC

Share logs here as well, might help update the integration guide.

Following are the major reasons why you'll fall into the voicemail application:

1 - FS failed to Dial to Kamailio, probably unable to reach Kamailio or syntax 
problem in the originate/bridge etc
2 - FS dialled to Kamailio but the route file is not properly setup to handle 
calls from FS and lookup() the user.
3 - Kamailio is setup correctly but the user is not online, or the lookup() 
don't have the user as FS required in uesrlocation table, or the end user 
doesn't accept the codecs.

I mentioned the mismatch in domain part in RURI in one of my previous emails 
looking at your  sip traces, you've already modified the packet but I still 
need to take a look at the sip captures to verify this.

Thanks,
Sammy




On Thu, Feb 11, 2016 at 12:28 PM, malik sherif 
<[email protected]<mailto:[email protected]>> wrote:

Hello Sammy,

I used both the gateway method and external, the result is the same it goes the 
voicemail. I enabled debug on FS an should I post my question to FS? I followed 
the steps that was in kamailio to integrate kamailio and FS to setup SBC and 
that way I posted on kamailio site.

Thanks

Abdul


________________________________
From: sr-users 
<[email protected]<mailto:[email protected]>>
 on behalf of SamyGo <[email protected]<mailto:[email protected]>>
Sent: Wednesday, February 10, 2016 10:23 PM

To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC

Hi Abdul,

Kindly share the whole FS console logs (enable sip debug inside the logs too) , 
can you modify the bridge statement as this:

<action application="bridge" data="sofia/external/[email protected]"/>

If you have saved your kamailio as a gateway then you can alternatively dial it 
as following:

<action application="bridge" data="sofia/gateway/GOOD_GATEWAY/$1"/>

Where GOOD_GATEWAY is the gateway name from an xml file. Here is how.

FreeSWITCH:~# cd /usr/local/freeswitch/conf/sip_profiles/external/

FreeSWITCH-A:~# vim kamailio.xml

Insert these Lines in this file:

<include>
  <gateway name="GOOD_GATEWAY">
  <param name="username" value="nothing"/>
  <param name="password" value="doesn't_matter"/>
  <param name="proxy" value="192.168.30.3"/>     <!--SET IP OF KAMAILIO HERE -->
  <param name="register" value="false"/>
  <param name="retry-seconds" value="10"/>
  <param name="caller-id-in-from" value="true"/>
  <param name="extension-in-contact" value="true"/>
  <param name="ping" value="25"/>
  <param name="inbound-late-negotiation" value="true"/>
  <param name="context" value="default"/>
  </gateway>
</include>

Also, if you don't use gateway approach can you make sure that from your FS the 
domain name 'AbdulKamailioSIP.com' resolves to IP of Kamailio Server.

I've a feeling that this email should be in Freeswitch mailing list, not in 
Kamailio's/

Regards,
Sammy



On Wed, Feb 10, 2016 at 5:00 PM, malik sherif 
<[email protected]<mailto:[email protected]>> wrote:

Hello,

I am using Kamailio and freeswitch to setup SBC but the I attempted to make a 
call it just goes to the voice mail.

Here is what freeswitch is displaying.

Thanks for your help in advance

Abdul



freeswitch@linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE] 
switch_channel.c:1055 New Channel sofia/internal/[email protected] 
[12f87c10-f3be-43ee-b038-f6647e5af373]
2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 
<102>->kb-102 in context public
2016-02-10 10:54:16.683337 [NOTICE] switch_ivr.c:1861 Transfer 
sofia/internal/[email protected] to XML[kb-102@default]
2016-02-10 10:54:16.683337 [INFO] mod_dialplan_xml.c:635 Processing 102 
<102>->kb-102 in context default
2016-02-10 10:54:16.683337 [NOTICE] switch_channel.c:1055 New Channel 
sofia/internal/[email protected] [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3]
2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup 
sofia/internal/[email protected] [CS_CONSUME_MEDIA] 
[NORMAL_TEMPORARY_FAILURE]
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 
(sofia/internal/[email protected]) Ended
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel 
sofia/internal/[email protected] [CS_DESTROY]
2016-02-10 10:54:18.183346 [INFO] mod_dptools.c:3244 Originate Failed.  Cause: 
NORMAL_TEMPORARY_FAILURE
2016-02-10 10:54:18.183346 [NOTICE] sofia_media.c:92 Pre-Answer 
sofia/internal/[email protected]!
2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel 
[sofia/internal/[email protected]] has been answered
2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup 
sofia/internal/[email protected] [CS_EXECUTE] [NORMAL_CLEARING]
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 
(sofia/internal/[email protected]) Ended
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel 
sofia/internal/[email protected] [CS_DESTROY]



Any idea as to how to implement this command on freeswitch dial plan, I am not 
sure what to use for gw1

<action application="bridge" 
data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/[email protected]<mailto:[email protected]>"/>





>From Freeswitch dial plan


<extension name="kbridge">
        <condition field="destination_number" expression="^kb-(.+)$">
                  <action application="set" data="proxy_media=true"/>
                  <action application="set" data="call_timeout=50"/>
                  <action application="set" data="continue_on_fail=true"/>
                  <action application="set" data="hangup_after_bridge=true"/>
                <action application="set" 
data="sip_invite_domain=AbdulkamailioSIP.com"/>
                  <action application="export" data="sip_contact_user=ufs"/>
                <action application="bridge" 
data="sofia/$${domain}/[email protected]"/>
                  <action application="answer"/>
                  <action application="voicemail" data="default ${domain_name} 
$1"/>
        </condition>
      </extension>






________________________________
From: sr-users 
<[email protected]<mailto:[email protected]>>
 on behalf of SamyGo <[email protected]<mailto:[email protected]>>
Sent: Friday, January 29, 2016 5:02 PM

To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC

Sorry for last email:
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
That is where you get 404 Not Found. What I see is that you're registering 
users with domain as AbdulKamailioSIP.com but when your FreeSwitch sends call 
to Kamailio the RURI becomes: INVITE 
sip:[email protected]<mailto:sip%[email protected]> SIP/2.0 Which is 
definitely not matching any User like: INVITE 
sip:[email protected] SIP/2.0 So, you need to go in your FS 
dialplan and make sure you set the proper Domains before sending call out, 
there are couple of ways to do this. 1 - Using FreeSWITCH to set FROM domain: 
https://wiki.freeswitch.org/wiki/Variable_sip_invite_domain 2 - Use custom SIP 
header from FS to contain a domain name, and in Kamailio set headers as you 
require; something like this: Attach a SIP Header in FS dialplan before sending 
call out to Kamailio, say X-USER-DOMAIN: AbdulKamailioSIP.com Next when I 
receive call in Kamailio.cfg I detect this header 
if(is_present_hf("X-USER-DOMAIN")) { $ru = "sip:" + $rU + "@" + 
$hdr(X-USER-DOMAIN); $td = $hdr(X-USER-DOMAIN); } In option 2 you must do it 
before executing record_route() functions, so possibly need to do this inside 
your FSINBOUND route. I prefer option 1. PS: Wireshark highlights any custom 
SIP headers in sky blue, that doesn't mean there is any error in there.

Regards,
Sammy


On Fri, Jan 29, 2016 at 11:47 AM, SamyGo 
<[email protected]<mailto:[email protected]>> wrote:
Hi Abdul,

This is where you are getting your 404 NOT Found from Kamailio:



On Thu, Jan 28, 2016 at 4:30 PM, malik sherif 
<[email protected]<mailto:[email protected]>> wrote:

I will also run the commands that suggested.


________________________________
From: sr-users 
<[email protected]<mailto:[email protected]>>
 on behalf of SamyGo <[email protected]<mailto:[email protected]>>
Sent: Thursday, January 28, 2016 6:08 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC

I believe Daniel is busy with FOSDEM ,


Abdul can you confirm that you're still getting this output in FS console:

2016-01-13 05:37:29.572184 [INFO] mod_dialplan_xml.c:635 Processing 7632689991 
<7632689991>->kb-7632689993 in context default
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING 
WARNING WARNING WARNING WARNING WARNING WARNING
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Open 
/usr/local/freeswitch/conf/vars.xml and change the default_password.
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 Once changed type 
'reloadxml' at the console.
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WARNING WARNING WARNING 
WARNING WARNING WARNING WARNING WARNING WARNING
2016-01-13 05:37:39.632245 [NOTICE] switch_channel.c:1055 New Channel 
sofia/internal/[email protected]<mailto:[email protected]> 
[d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup 
sofia/internal/[email protected]<mailto:[email protected]> [CS_ROUTING] 
[UNALLOCATED_NUMBER]

Please paste your complete dialplan here as well, though this clearly states 
that the number it tried to dial is not registered or unable to dial to.
please paste out the content of the following command just before dialing:
fs_cli> show registrations

Also, it will help you find out useful info about why it shows you UNALLOCATED 
NUMBER if you enable the sofia sip debug by using the following command.
fs_cli> sofia global siptrace on

Once you execute the above command make a call to destination and see what 
FreeeSWITCH is trying to do.

Thanks,
Sammy.

On Thu, Jan 28, 2016 at 11:23 AM, malik sherif 
<[email protected]<mailto:[email protected]>> wrote:


Any hint?

________________________________
From: sr-users 
<[email protected]<mailto:[email protected]>>
 on behalf of malik sherif <[email protected]<mailto:[email protected]>>
Sent: Tuesday, January 26, 2016 11:35 PM
To: Kamailio (SER) - Users Mailing List; 
[email protected]<mailto:[email protected]>

Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC


Thanks again and here is the pcap file.

Thanks

Abdul


________________________________
From: Daniel-Constantin Mierla <[email protected]<mailto:[email protected]>>
Sent: Friday, January 22, 2016 8:46 AM
To: malik sherif; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC

Can you attach the pcap file - copy&paste inline makes it imposible to read and 
digest it with a traffic analyzer (e.g., wireshark).

Cheers,
Daniel

On 21/01/16 18:31, malik sherif wrote:



________________________________
From: sr-users 
<[email protected]><mailto:[email protected]>
 on behalf of malik sherif <[email protected]><mailto:[email protected]>
Sent: Wednesday, January 20, 2016 9:55 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio and freeswitch integration for SBC


Copy and paste part of tcmdump and highlighted the 404. 10.22.52.2 is the 
server IP address

Thanks again

Abdul

[http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>


--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu

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[email protected]<mailto:[email protected]>
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