[SR-Users] Fw: Kamailio and freeswitch integration for SBC
SamyGo
govoiper at gmail.com
Thu Feb 11 18:41:46 CET 2016
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
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>
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