[SR-Users] Fw: Kamailio and freeswitch integration for SBC
SamyGo
govoiper at gmail.com
Fri Feb 12 00:50:57 CET 2016
Use pastebin.com or something ?
On Feb 11, 2016 18:32, "malik sherif" <asherif74 at hotmail.com> 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/
> 102 at newkama.AbdulKamailioSIP.com Restore previous codec PCMU:0.
> 2016-02-11 11:38:42.549341 [DEBUG] mod_voicemail.c:2806 Deliver VM to
> 101 at 10.22.52.2
> 2016-02-11 11:38:42.669308 [DEBUG] mod_voicemail.c:1923 Update MWI:
> Processing for 101 at 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 at 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 at 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 at 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 at 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 at newkama.AbdulKamailioSIP.com) Callstate Change ACTIVE
> -> HANGUP
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:737
> (sofia/internal/102 at newkama.AbdulKamailioSIP.com) State HANGUP
> 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:407 sofia/internal/
> 102 at 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 at 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 at 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 at 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 at 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 at newkama.AbdulKamailioSIP.com [BREAK]
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/102 at 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 at newkama.AbdulKamailioSIP.com) State REPORTING
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/102 at 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 at 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 at 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 at newkama.AbdulKamailioSIP.com [BREAK]
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_session.c:1623 Session 7
> (sofia/internal/102 at 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 at newkama.AbdulKamailioSIP.com) Ended
> 2016-02-11 11:38:42.669308 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/102 at newkama.AbdulKamailioSIP.com [CS_DESTROY]
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:626
> (sofia/internal/102 at 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 at newkama.AbdulKamailioSIP.com) State DESTROY
> 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/
> 102 at newkama.AbdulKamailioSIP.com SOFIA DESTROY
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111
> sofia/internal/102 at newkama.AbdulKamailioSIP.com Standard DESTROY
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:636
> (sofia/internal/102 at newkama.AbdulKamailioSIP.com) State DESTROY going to
> sleep
>
>
>
> ------------------------------
> *From:* sr-users <sr-users-bounces at lists.sip-router.org> on behalf of
> SamyGo <govoiper at 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 at 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 at lists.sip-router.org> on behalf of
>> SamyGo <govoiper at 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 at 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 at linux-ix64> 2016-02-10 10:54:16.663387 [NOTICE]
>>> switch_channel.c:1055 New Channel sofia/internal/102 at 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 at AbdulKamailioSIP.com to XML[kb-102 at 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 at AbdulkamailioSIP.com
>>> [0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3]
>>> 2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup
>>> sofia/internal/102 at 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 at AbdulkamailioSIP.com) Ended
>>> 2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close
>>> Channel sofia/internal/102 at 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 at AbdulKamailioSIP.com!
>>> 2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel
>>> [sofia/internal/102 at AbdulKamailioSIP.com] has been answered
>>> 2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup
>>> sofia/internal/102 at AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1
>>> (sofia/internal/102 at AbdulKamailioSIP.com) Ended
>>> 2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close
>>> Channel sofia/internal/102 at 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/$1 at domain.org"/>
>>>
>>>
>>>
>>>
>>>
>>> From Freeswitch dial plan
>>>
>>>
>>> <extension name="kbridge">
>>> <condition field="destination_number" expression="^kb-(.+)$">
>>> <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 at AbdulkamailioSIP.com"/>
>>> <action application="answer"/>
>>> <action application="voicemail" data="default
>>> ${domain_name} $1"/>
>>> </condition>
>>> </extension>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------
>>> *From:* sr-users <sr-users-bounces at lists.sip-router.org> on behalf of
>>> SamyGo <govoiper at 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 at 10.22.52.2 <sip%3A7632689993 at 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 at 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 at hotmail.com>
>>>> wrote:
>>>>
>>>>> I will also run the commands that suggested.
>>>>>
>>>>>
>>>>> ------------------------------
>>>>> *From:* sr-users <sr-users-bounces at lists.sip-router.org> on behalf of
>>>>> SamyGo <govoiper at 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 at 10.22.52.2
>>>>> [d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
>>>>> 2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/
>>>>> 7632689993 at 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 at hotmail.com>
>>>>> wrote:
>>>>>
>>>>>>
>>>>>> Any hint?
>>>>>>
>>>>>> ------------------------------
>>>>>> *From:* sr-users <sr-users-bounces at lists.sip-router.org> on behalf
>>>>>> of malik sherif <asherif74 at hotmail.com>
>>>>>> *Sent:* Tuesday, January 26, 2016 11:35 PM
>>>>>> *To:* Kamailio (SER) - Users Mailing List; miconda at 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 at 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 at lists.sip-router.org>
>>>>>> <sr-users-bounces at lists.sip-router.org> on behalf of malik sherif
>>>>>> <asherif74 at hotmail.com> <asherif74 at 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
>>>>>>
>>>>>>
>>>>>> <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>>>>> Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>>>>> list
>>>>>> sr-users at lists.sip-router.org
>>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>>> sr-users at lists.sip-router.org
>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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