11 Feb
2016
11 Feb
'16
12:41 p.m.
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(a)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(a)lists.sip-router.org> on behalf of
SamyGo <govoiper(a)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(a)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(a)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(a)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(a)AbdulkamailioSIP.com
[0c6c8dda-34fc-45a0-a6a2-8e82ff3a9be3]
2016-02-10 10:54:18.183346 [NOTICE] sofia.c:7539 Hangup
sofia/internal/102(a)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(a)AbdulkamailioSIP.com) Ended
2016-02-10 10:54:18.183346 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal/102(a)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(a)AbdulKamailioSIP.com!
2016-02-10 10:54:18.183346 [NOTICE] mod_dptools.c:1268 Channel
[sofia/internal/102(a)AbdulKamailioSIP.com] has been answered
2016-02-10 10:54:32.043345 [NOTICE] sofia.c:952 Hangup
sofia/internal/102(a)AbdulKamailioSIP.com [CS_EXECUTE] [NORMAL_CLEARING]
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1641 Session 1
(sofia/internal/102(a)AbdulKamailioSIP.com) Ended
2016-02-10 10:54:32.063338 [NOTICE] switch_core_session.c:1645 Close
Channel sofia/internal/102(a)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/$1(a)domain.org"/>
From Freeswitch dial plan
<extension name="kbridge">
<condition field="destination_number"
expression="^kb-(.+)$">
<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(a)AbdulkamailioSIP.com"/>
<action application="answer"/>
<action application="voicemail" data="default
${domain_name} $1"/>
</condition>
</extension>
------------------------------
*From:* sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of
SamyGo <govoiper(a)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%3A7632689993(a)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(a)gmail.com> wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)hotmail.com>
wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I will also run the commands that suggested.
------------------------------
*From:* sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of
SamyGo <govoiper(a)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(a)10.22.52.2
[d52b6ef9-c4f6-4edf-aff9-8a8da3761788]
2016-01-13 05:37:39.632245 [NOTICE] sofia.c:7539 Hangup sofia/internal/
7632689993(a)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(a)hotmail.com>
wrote:
>
> Any hint?
>
> ------------------------------
> *From:* sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of
> malik sherif <asherif74(a)hotmail.com>
> *Sent:* Tuesday, January 26, 2016 11:35 PM
> *To:* Kamailio (SER) - Users Mailing List; miconda(a)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(a)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(a)lists.sip-router.org>
> <sr-users-bounces(a)lists.sip-router.org> on behalf of malik sherif
> <asherif74(a)hotmail.com> <asherif74(a)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(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
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sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users