22 Feb
2014
22 Feb
'14
2:46 a.m.
Hello, I have a problem that my calls get rejected by he pstn gateway as my gateway is
expecting a DID number instead of a 4 digit extension as the 'from user". Let me
explain my situation:
1. I have set callfwd_busy in my user preference table.
2. In my failure route I check to see if callfwd_busy has been set, if it has then I
execute the following script:
if (t_check_status("486|408")) {
$avp(dst_number) = $rU;
if (avp_db_load("$avp(dst_number)",
"$avp(callfwd_busy)")) {
xlog("LOG: avp(callfwd_busy)=$avp(callfwd_busy),
avp(dst_number)=$avp(dst_number)\n");
}
xlog("LOG: Failure route with $rU $avp(dst_number)
\n");
$rU = $avp(callfwd_busy);
$ru = "sip:" + $rU + "@" +
$sel(cfg_get.pstn.gw_ip);
route(RELAY);
exit;
}
When I call from extension 7004 to ext 7002 ( this has callfwd_busy set to a DID number)
the trace looks as follows:
U 172.10.30.15:5080 -> 172.10.30.15:5060
SIP/2.0 486 Busy Here.
Via: SIP/2.0/UDP 54.200.xx.yy:5060;branch=z9hG4bKee75.ab791c5.0;received=172.10.30.15.
Via: SIP/2.0/UDP
192.168.1.8:5060;received=71.252.219.63;branch=z9hG4bK209939211;rport=1025.
From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367.
To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40.
Call-ID: 876294364-5060-74(a)BJC.BGI.B.I.
CSeq: 731 INVITE.
Server: Asterisk PBX 1.8.17.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
U 172.10.30.15:5060 -> 172.10.30.15:5080
INVITE sip:18455981234@64.136.xx.yy:5060 SIP/2.0.
Record-Route: <sip:54.200.xx.xx;lr=on;ftag=1830267367;nat=yes>.
Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1.
Via: SIP/2.0/UDP
192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025.
From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367.
To: <sip:7002@ajfmc1.myDomain.net>.
Call-ID: 876294364-5060-74(a)BJC.BGI.B.I.
CSeq: 731 INVITE.
Contact: "7004" <sip:7004@71.252.xx.yy:1025>.
Max-Forwards: 16.
User-Agent: Grandstream HT701 1.0.4.8.
Privacy: none.
P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>.
Supported: replaces, path, timer, eventlist.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE.
Content-Type: application/sdp.
Accept: application/sdp, application/dtmf-relay.
Content-Length: 453.
.
v=0.
o=7004 8000 8000 IN IP4 172.10.30.1554.200.xx.xx.
s=SIP Call.
c=IN IP4 172.10.30.1554.200.xx.xx.
t=0 0.
m=audio 6248262482 RTP/AVP 0 18 4 8 2 97 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16,32-36,54.
a=nortpproxy:yes.
a=nortpproxy:yes.
#
U 172.10.30.15:5080 -> 172.10.30.15:5060
SIP/2.0 503 Unavailable.
Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1;received=172.10.30.15.
Via: SIP/2.0/UDP
192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025.
From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367.
To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40.
Call-ID: 876294364-5060-74(a)BJC.BGI.B.I.
CSeq: 731 INVITE.
Server: Asterisk PBX 1.8.17.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
It looks like my gateway is expecting to see a DID number in the from header. Can someone
let me know how I can dot this? Thank you.
Arun
22 Feb
22 Feb
2:49 a.m.
On 02/21/2014 07:46 PM, arun Jayaprakash wrote:
It looks like my gateway is expecting to see a DID
number in the from
header. Can someone let me know how I can dot this? Thank you.
The uac_replace_from() function exported by the 'uac' module can do this:
http://kamailio.org/docs/modules/4.1.x/modules/uac.html#uac.f.uac_replace_f…
However, you should carefully give some consideration to whether you
really want to change the value of the From header. It is quite
inadvisable, and is not something proxies are formally permitted to do.
The UAC module gets around this requirement through some rather
complicated spoofing.
You might consider instead reconfiguring your UAC to send different From
values, most likely by changing 'caller ID' settings in Asterisk, in
your case. For instance, there is the 'callerid' option for sip,conf peers.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
3:16 a.m.
Alex, thank you for your pointers. I will work with Asterisk to see how I can change the
caller ID instead of messing with the UAC module. Thanks again.
Regards,
Arun
________________________________
From: Alex Balashov <abalashov(a)evaristesys.com>
To: arun Jayaprakash <jayaprakasharun(a)yahoo.com>om>; Kamailio (SER) - Users Mailing
List <sr-users(a)lists.sip-router.org>
Sent: Friday, February 21, 2014 6:49 PM
Subject: Re: [SR-Users] Change from_uri
On 02/21/2014 07:46 PM, arun Jayaprakash wrote:
It looks like my gateway is expecting to see a DID
number in the from
header. Can someone let me know how I can dot this? Thank you.
The uac_replace_from() function exported by the 'uac' module can do this:
http://kamailio.org/docs/modules/4.1.x/modules/uac.html#uac.f.uac_replace_f…
However, you should carefully give some consideration to whether you really want to change
the value of the From header. It is quite inadvisable, and is not something proxies are
formally permitted to do. The UAC module gets around this requirement through some rather
complicated spoofing.
You might consider instead reconfiguring your UAC to send different From values, most
likely by changing 'caller ID' settings in Asterisk, in your case. For instance,
there is the 'callerid' option for sip,conf peers.
-- Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
24 Feb
24 Feb
11 a.m.
On Saturday 22 February 2014 02:16:01 arun Jayaprakash wrote:
Alex, thank you for your pointers. I will work with
Asterisk to see how I
can change the caller ID instead of messing with the UAC module. Thanks
again.
You might want to check what happens when you fix the header:
P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>
by simply deleting it and adding a correct one.
--
POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/ - http://www.sipo.nl/
K.v.K. Eindhoven 17097024
11:01 a.m.
True! As most industrial equipment out there prefers PAI > RPID > From.
Daniel Tryba <daniel(a)pocos.nl> wrote:
On Saturday 22 February 2014 02:16:01 arun Jayaprakash
wrote:
--
Sent from my mobile, and thus lacking in the refinement one might expect from a fully
fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com
Alex, thank you for your pointers. I will work
with Asterisk to see
how I
can change the caller ID instead of messing with
the UAC module.
Thanks
again.
You might want to check what happens when you fix the header:
P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>
by simply deleting it and adding a correct one.
3924
days inactive
3926
days old
4 comments
3 participants
participants (3)
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Alex Balashov -
arun Jayaprakash -
Daniel Tryba