10 Feb
2016
10 Feb
'16
5:23 p.m.
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:
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
_______________________________________________
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