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/102@newkama.AbdulKamailioSIP.com Restore previous codec PCMU:0.
2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to 101@10.22.52.2
2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI: Processing for 101@10.22.52.2 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 101@10.22.52.2
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/102@newkama.AbdulKamailioSIP.com 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/102@newkama.AbdulKamailioSIP.com) State EXECUTE going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Running State Change CS_HANGUP
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Callstate Change ACTIVE -> HANGUP
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State HANGUP
2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/102@newkama.AbdulKamailioSIP.com 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/102@newkama.AbdulKamailioSIP.com hanging up, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:60 sofia/internal/102@newkama.AbdulKamailioSIP.com Standard HANGUP, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State HANGUP going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State Change CS_HANGUP -> CS_REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com [BREAK]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Running State Change CS_REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State REPORTING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104 sofia/internal/102@newkama.AbdulKamailioSIP.com Standard REPORTING, cause: NORMAL_CLEARING
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State REPORTING going to sleep
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State Change CS_REPORTING -> CS_DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/102@newkama.AbdulKamailioSIP.com [BREAK]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Locked, Waiting on external entities
2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1641 Session 7 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Ended
2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@newkama.AbdulKamailioSIP.com [CS_DESTROY]
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/102@newkama.AbdulKamailioSIP.com) Running State Change CS_DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State DESTROY
2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/102@newkama.AbdulKamailioSIP.com SOFIA DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111 sofia/internal/102@newkama.AbdulKamailioSIP.com Standard DESTROY
2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/102@newkama.AbdulKamailioSIP.com) State DESTROY going to sleep

---

From: sr-users <sr-users-bounces@lists.sip-router.org> on behalf of SamyGo <govoiper@gmail.com>
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 <asherif74@hotmail.com> 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 <sr-users-bounces@lists.sip-router.org> on behalf of SamyGo <govoiper@gmail.com>
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/$1@AbdulkamailioSIP.com"/>

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 <asherif74@hotmail.com> 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/102@AbdulKamailioSIP.com [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/102@AbdulKamailioSIP.com 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/102@AbdulkamailioSIP.com [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3]
2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup sofia/internal/102@AbdulkamailioSIP.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1641 Session 2 (sofia/internal/102@AbdulkamailioSIP.com) Ended
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@AbdulkamailioSIP.com [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/102@AbdulKamailioSIP.com!
2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel [sofia/internal/102@AbdulKamailioSIP.com] has been answered
2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup sofia/internal/102@AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING]
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1 (sofia/internal/102@AbdulKamailioSIP.com) Ended
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/102@AbdulKamailioSIP.com [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/$1@domain.org"/>

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}/$1@AbdulkamailioSIP.com"/>
                  <action application="answer"/>
                  <action application="voicemail" data="default ${domain_name} $1"/>
        </condition>
      </extension>

---

From: sr-users <sr-users-bounces@lists.sip-router.org> on behalf of SamyGo <govoiper@gmail.com>
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:7632689993@10.22.52.2 SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993@AbdulKamailioSIP.com 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 <govoiper@gmail.com> 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 <asherif74@hotmail.com> wrote:

I will also run the commands that suggested.

---

From: sr-users <sr-users-bounces@lists.sip-router.org> on behalf of SamyGo <govoiper@gmail.com>
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/7632689993@10.22.52.2 [d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/7632689993@10.22.52.2 [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 <asherif74@hotmail.com> wrote:

Any hint?

---

From: sr-users <sr-users-bounces@lists.sip-router.org> on behalf of malik sherif <asherif74@hotmail.com>
Sent: Tuesday, January 26, 2016 11:35 PM
To: Kamailio (SER) - Users Mailing List; miconda@gmail.com

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

 

Thanks again and here is the pcap file.

Thanks

Abdul

---

From: Daniel-Constantin Mierla <miconda@gmail.com>
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 <sr-users-bounces@lists.sip-router.org> on behalf of malik sherif <asherif74@hotmail.com>
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

-- 
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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