29 Jan
2016
29 Jan
'16
12:02 p.m.
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:
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
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