[SR-Users] Fw: Kamailio and freeswitch integration for SBC
malik sherif
asherif74 at hotmail.com
Thu Feb 11 18:44:28 CET 2016
Thanks Sammy again,
I just post the log debug.
Thanks
Abdul
________________________________
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<mailto: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<mailto:sr-users-bounces at lists.sip-router.org>> on behalf of SamyGo <govoiper at gmail.com<mailto: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 at 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<mailto: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/$1 at domain.org<mailto:1 at domain.org>"/>
>From Freeswitch dial plan
<extension name="kbridge">
<condition field="destination_number" expression="^kb-(.+)$">
<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<mailto:sr-users-bounces at lists.sip-router.org>> on behalf of SamyGo <govoiper at gmail.com<mailto: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<mailto:sip%3A7632689993 at 10.22.52.2> SIP/2.0 Which is definitely not matching any User like: INVITE sip:7632689993 at 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<mailto: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<mailto:asherif74 at hotmail.com>> wrote:
I will also run the commands that suggested.
________________________________
From: sr-users <sr-users-bounces at lists.sip-router.org<mailto:sr-users-bounces at lists.sip-router.org>> on behalf of SamyGo <govoiper at gmail.com<mailto: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<mailto: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<mailto: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<mailto:asherif74 at hotmail.com>> wrote:
Any hint?
________________________________
From: sr-users <sr-users-bounces at lists.sip-router.org<mailto:sr-users-bounces at lists.sip-router.org>> on behalf of malik sherif <asherif74 at hotmail.com<mailto:asherif74 at hotmail.com>>
Sent: Tuesday, January 26, 2016 11:35 PM
To: Kamailio (SER) - Users Mailing List; miconda at gmail.com<mailto: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<mailto: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><mailto:sr-users-bounces at lists.sip-router.org> on behalf of malik sherif <asherif74 at hotmail.com><mailto: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/_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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