[SR-Users] Fw: Kamailio and freeswitch integration for SBC
SamyGo
govoiper at gmail.com
Wed Feb 10 23:23:22 CET 2016
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
>
>
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