Hello list.
I have reached a point where i need some advices and a little help.
I'm using mediaproxy to solve the NAT'd client "problems" this module works
fine, and solves all the scenarios that i'm interested but for i noticed i
alway have to enable the mediaproxy, even for the calls between two clients
with public.ip, this is a problem for me, i would expected that the media in
this case goes directly between the endpoints.
Mi first question is :
Is this behaivor normal? Is that the way the mediaproxy works?
Second Question.
If i use nathelper, would i have the same problem?. Among the nathelper
users do you have any similar experience. Does anyone knows if Nathelper
solve this issue?.
I really, really hope that someone can help me with these questions.
Thanks in advance.
Best Regards
Ricardo
have you run ./configure first?
On Tuesday 28 September 2004 20:46, Ricardo Martinez wrote:
> Hello.
> I'm trying to compile the RTP proxy. I downloades the last version
> from the CVS with :
>
> [root@sipproxy root]# cvs
> -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy
>
> cvs server: Updating rtpproxy
> U rtpproxy/Makefile.am
> U rtpproxy/Makefile.in
> U rtpproxy/README
> U rtpproxy/README.remote
> U rtpproxy/aclocal.m4
> U rtpproxy/config.h.in
> U rtpproxy/configure
> U rtpproxy/configure.ac
> U rtpproxy/depcomp
> U rtpproxy/install-sh
> U rtpproxy/main.c
> U rtpproxy/missing
> U rtpproxy/mkinstalldirs
> U rtpproxy/myqueue.h
> [root@sipproxy root]#
>
> Then i try to compile the program
>
> [root@sipproxy rtpproxy]# make
> make all-am
> make[1]: Entering directory `/root/rtpproxy'
> source='main.c' object='main.o' libtool=no \
> depfile='.deps/main.Po' tmpdepfile='.deps/main.TPo' \
> depmode=gcc /bin/sh ./depcomp \
> gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -c `test -f 'main.c' || echo
> './'`main.c
> main.c: In function `main':
> main.c:1152: structure has no member named `ss_family'
> make[1]: *** [main.o] Error 1
> make[1]: Leaving directory `/root/rtpproxy'
> make: *** [all] Error 2
> [root@sipproxy rtpproxy]#
>
> I'm using Red Hat Linux 7.3
>
> Any ideas?
>
>
> Ricardo Martinez.
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Hi all.
I'm using ser 0.8.99-dev6 with serweb. I'm not sure which version of serweb
this is but I checked it out of berlios today with this command:
cvs -z3 -d:pserver:anonymous@cvs.serweb.berlios.de:/cvsroot/serweb co iptel
Anyhow, my click-to-dial feature is not fully functional. When I click on an
entry in the phonebook it rings my extension as it should, but when I go off
hook the number I'm calling never rings.
Here are the click-to-dial settings from <serweb>/config/config.php
$config->ctd_target="sip:699@68.90.50.100";
$config->ctd_uri="sip:699@68.90.50.100";
$config->ctd_from="sip:699@mycompany.com";
$config->ctd_outbound_proxy="";
Account 699 does not actually exist in my ser/subscriber table in MySQL. I'm
very unclear on what these parameters should be set to.
Also here is the ngrep output from my click-to-dial attempt. As you can see
about half way down there is a REFER message but it seems to point to my
Asterisk voice mail server (vm01.mycompany.com). Shouldn't this point to the
person in my phone book that I'm calling?
In this call sequence I called sip:1002@mycompany.com from
sip:1000@mycompany.com by clicking on the 1002 phonebook entry.
###
U 68.90.50.100:5060 -> 12.3.4.10:5060
INVITE sip:1001@12.3.4.10;user=phone SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0.
To: <sip:1001@mycompany.com>.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
CSeq: 1 INVITE.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 131.
Contact: <sip:caller@68.90.50.100:5060>.
Reject-Contact: *;automata="YES".
Content-Type: application/sdp.
.
v=0.
o=click-to-dial 0 0 IN IP4 0.0.0.0.
s=session.
c=IN IP4 0.0.0.0.
b=CT:1000.
t=0 0.
m=audio 9 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.
#
U 12.3.4.10:5060 -> 68.90.50.100:5060
SIP/2.0 100 trying.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
To: <sip:1001@mycompany.com>.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 1 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Warning: 399 12.3.4.10 "detected NAT type is full cone".
Content-Length: 0.
.
#
U 12.3.4.10:5060 -> 68.90.50.100:5060
SIP/2.0 180 ringing.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 1 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Warning: 399 12.3.4.10 "detected NAT type is full cone".
Content-Length: 0.
.
##
U 12.3.4.10:5060 -> 68.90.50.100:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.95425785.0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 1 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Warning: 399 12.3.4.10 "detected NAT type is full cone".
Contact: <sip:1001@12.3.4.10;user=phone>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 161.
.
v=0.
o=1001 8000 8000 IN IP4 12.3.4.10.
s=SIP Call.
c=IN IP4 12.3.4.10.
t=0 0.
m=audio 5004 RTP/AVP 0.
a=recvonly.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
#
U 68.90.50.100:5060 -> 12.3.4.10:5060
ACK sip:1001@12.3.4.10;user=phone SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
Via: SIP/2.0/UDP 68.90.50.100;branch=0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK25c5.85425785.0.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
Call-ID: 415a15814acb0.fifouacctd.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
CSeq: 1 ACK.
Content-Length: 0.
.
#
U 68.90.50.100:5060 -> 68.84.242.201:5060
REFER sip:1001@vm01.mycompany.com:5060;user=phone SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
Via: SIP/2.0/UDP 68.90.50.100;branch=0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bKf4c5.16082cd7.0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
CSeq: 2 REFER.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 0.
Contact: <sip:caller@68.90.50.100:5060>.
Referred-By: <sip:699@mycompany.com>.
Refer-To: sip:1002@mycompany.com.
.
#
U 68.84.242.201:5060 -> 68.90.50.100:5060
SIP/2.0 202 Accepted.
Via: SIP/2.0/UDP 68.90.50.100;branch=0;received=68.90.50.100;rport=5060.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bKf4c5.16082cd7.0.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 2 REFER.
User-Agent: VoiceMail.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: .
Content-Length: 0.
.
##
U 68.90.50.100:5060 -> 68.84.242.201:5060
BYE sip:1001@vm01.mycompany.com:5060;user=phone SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
Via: SIP/2.0/UDP 68.90.50.100;branch=0.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK05c5.3f0aaa06.0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
CSeq: 3 BYE.
Call-ID: 415a15814acb0.fifouacctd.
Content-Length: 0.
Contact: <sip:caller@68.90.50.100:5060>.
.
#
U 68.84.242.201:5060 -> 68.90.50.100:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.90.50.100;branch=0;received=68.90.50.100;rport=5060.
Via: SIP/2.0/UDP 68.90.50.100;branch=z9hG4bK05c5.3f0aaa06.0.
Record-Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
From: <sip:699@mycompany.com>;tag=415a15814acb0.
To: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 3 BYE.
User-Agent: VoiceMail.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: .
Content-Length: 0.
.
##########
U 12.3.4.10:5060 -> 68.90.50.100:5060
BYE sip:caller@68.90.50.100:5060 SIP/2.0.
Via: SIP/2.0/UDP 12.3.4.10;branch=z9hG4bKbb2f53e68cca3b12.
Route: <sip:68.90.50.100;ftag=415a15814acb0;lr=on>.
From: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
To: <sip:699@mycompany.com>;tag=415a15814acb0.
Contact: <sip:1001@12.3.4.10;user=phone>.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 27932 BYE.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 68.90.50.100:5060 -> 12.3.4.10:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 12.3.4.10;branch=z9hG4bKbb2f53e68cca3b12.
From: <sip:1001@mycompany.com>;tag=7a9a058a857c4aba.
To: <sip:699@mycompany.com>;tag=415a15814acb0.
Call-ID: 415a15814acb0.fifouacctd.
CSeq: 27932 BYE.
Content-Length: 0.
Warning: 392 68.90.50.100:5060 "Noisy feedback tells: pid=26213
req_src_ip=68.90.50.100 req_src_port=5060 in_uri=sip:caller@68.90.50.100:5060
out_uri=sip:caller@68.90.50.100:5060 via_cnt==2".
.
Any ideas why 1002 never rings?
Regards,
Paul
__________________________________
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Hi Michael,
I added the xlog as you told me and got the following results.
When I send sipsak command,the log messages on the console showed as
following
_________________________________________________________________
[root@sipserv ser]# ser start
Listening on
127.0.0.1 [127.0.0.1]:5060
134.190.64.164 [134.190.64.164]:5060
Aliases: sipserv:5060 localhost:5060 134.190.64.164:*
sipserv.eplgroup.tara.ca:* eplgroup.tara.ca:*
stateless - initializing
Maxfwd module- initializing
[root@sipserv ser]# 0(7068) mod_init(): Database connection opened
successfuly
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
0(0) INFO: udp_init: SO_RCVBUF is initially 110592
0(0) INFO: udp_init: SO_RCVBUF is finally 221184
9(0) INFO: fifo process starting: 7094
9(7094) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo...
5(7082) Begin route
5(7082) Got a OPTIONS (URI = 4500(a)eplgroup.tara.ca) from 134.190.64.164
5(7082) Authenticating Contact (sip:sipsak@sipserv.eplgroup.tara.ca:32839)
_________________________________________________________________
The ngrep output is as followed:
________________________________________________________________
[root@sipserv ngrep-1.42]# ./ngrep -n 5060 -d eth0 4500
interface: eth0 (134.190.64.160/255.255.255.224)
match: 4500
############################################
U 134.190.64.171:5060 -> 134.190.64.164:5060 REGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.171>..Call-ID:70000-5647a0b0@134.1
90.64.171..CSeq:2 REGISTER..Expires: 7200..User-Agent:Mitel-5055-SIP-Phone
2.0.1.23 08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
I 134.190.64.164 -> 134.190.64.171
3:10....E.......<.....@...@......p.HREGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.17
1>..Call-ID:70000-5647a0b0@134.190.64.171..CSeq:2 REGISTER..Expires:
7200..User-Agent:Mitel-5055-SIP-Phone 2.0.1.23
08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
U 134.190.64.171:5060 -> 134.190.64.164:5060 REGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.171>..Call-ID:70000-5647a0b0@134.1
90.64.171..CSeq:2 REGISTER..Expires: 7200..User-Agent:Mitel-5055-SIP-Phone
2.0.1.23 08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
I 134.190.64.164 -> 134.190.64.171 3:10
....E.......<.....@...@......p.HREGISTER sip:134.190.64.164
SIP/2.0..Via:SI P/2.0/UDP 134.190.64.171:5060..From:"4500"
sip:4500@134.190.64.164>;tag=7-
366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.17
1>..Call-ID:70000-5647a0b0@134.190.64.171..CSeq:2 REGISTER..Expires:
7200..
User-Agent:Mitel-5055-SIP-Phone 2.0.1.23 8000F0E8F03..Max-Forwards:70..Con
tent-Length:0....
#
U 134.190.64.171:5060 -> 134.190.64.164:5060 REGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.171>..Call-ID:70000-5647a0b0@134.1
90.64.171..CSeq:2 REGISTER..Expires: 7200..User-Agent:Mitel-5055-SIP-Phone
2.0.1.23 08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
I 134.190.64.164 -> 134.190.64.171 3:10
....E.......<.....@...@......p.HREGISTER sip:134.190.64.164
SIP/2.0..Via:SI
P/2.0/UDP 134.190.64.171:5060..From:"4500"
sip:4500@134.190.64.164>;tag=7-
366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.17
1>..Call-ID:70000-5647a0b0@134.190.64.171..CSeq:2 REGISTER..Expires:
7200.. User-Agent:Mitel-5055-SIP-Phone 2.0.1.23
08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
U 134.190.64.171:5060 -> 134.190.64.164:5060 REGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.171>..Call-ID:70000-5647a0b0@134.1
90.64.171..CSeq:2 REGISTER..Expires: 7200..User-Agent:Mitel-5055-SIP-Phone
2.0.1.23 08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
I 134.190.64.164 -> 134.190.64.171 3:10
....E.......<.....@...@......p.HREGISTER sip:134.190.64.164
SIP/2.0..Via:SI P/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-
366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.17
1>..Call-ID:70000-5647a0b0@134.190.64.171..CSeq:2 REGISTER..Expires:
7200.. User-Agent:Mitel-5055-SIP-Phone 2.0.1.23
08000F0E8F03..Max-Forwards:70..Content-Length:0....
###
U 134.190.64.171:5060 -> 134.190.64.164:5060 REGISTER sip:134.190.64.164
SIP/2.0..Via:SIP/2.0/UDP 134.190.64.171:5060..From:"4500"
<sip:4500@134.190.64.164>;tag=7-366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.171>..Call-ID:70000-5647a0b0@134.1
90.64.171..CSeq:2 REGISTER..Expires: 7200..User-Agent:Mitel-5055-SIP-Phone
2.0.1.23 08000F0E8F03..Max-Forwards:70..Content-Length:0....
#
I 134.190.64.164 -> 134.190.64.171 3:10
....E.......<.....@...@......p.HREGISTER sip:134.190.64.164
SIP/2.0..Via:SI
P/2.0/UDP 134.190.64.171:5060..From:"4500"
sip:4500@134.190.64.164>;tag=7-
366-5f947b49..To:<sip:4500@134.190.64.164>..Contact:<sip:4500@134.190.64.17
1>..Call-ID:70000-5647a0b0@134.190.64.171..CSeq:2 REGISTER..Expires:
7200..
User-Agent:Mitel-5055-SIP-Phone 2.0.1.23
08000F0E8F03..Max-Forwards:70..Content-Length:0....
##############
_______________________________________________________________
Thanks for your help and time,
Yilan
> In your next post send the actual ngrep output and any log messages from
> the
> console or /var/log/messages... Basically what is happening is that the
> URI
> isn't matching "myself". When you start SER it will tell you what it
> binds
> to. Include that too next time.
>
>
> Try this:
>
> xlog("L_INFO", "Got a %rm (URI = %ru) from %is");
> if(uri == myself)
> {
> xlog("L_INFO", "Authenticating Contact (%ct)");
>
> # Make sure they are a valid user on our proxy
> if (!www_authorize("yourdomain.com", "location"))
> {
> www_challenge("yourdomain.com", "0");
> break;
> };
>
> save("location");
> xlog("L_INFO", "Registered Contact (%ct)");
> break;
> };
>
> ----------------------------------------
>
> Michael Shuler, C.E.O.
> BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
> 682 High Point Lane
> East Peoria, IL 61611
> Office: (217) 585-0357
> Cell: (309) 657-6365
> Fax: (309) 213-3500
> E-Mail: mike(a)bwsys.net
> Customer Service: (877) 976-0711
>
>> -----Original Message-----
>> From: yilan(a)cs.dal.ca [mailto:yilan@cs.dal.ca]
>> Sent: Tuesday, September 28, 2004 9:30 AM
>> To: Michael Shuler
>> Cc: yilan(a)cs.dal.ca; serusers(a)lists.iptel.org
>> Subject: RE: [Serusers] Ser Register Problem
>>
>>
>> Hi Michael,
>>
>> Thank you for your reply. I really appreciate it.
>>
>> 1 - Check out xlog() so you can print out the URI, Contact, etc.
>>
>> I add xlog () in ser.cfg.Then for the sipsak command "sipsak -vv -s
>> sip:4500@eplgroup.tara.ca", it shows the following
>> 5(6374) not found:time [Tue Sep 28 11:14:48 2004] method
>> <OPTIONS> r-uri
>> <4500(a)eplgroup.tara.ca>
>> I guess sipsak doesn't send register method here.
>>
>> But for the phone that is going to register for the ser, nothing shows
>> although I can see the phone sends register packet from ngrep.
>>
>> 2 - Are you beheind NAT?
>> NO.
>>
>> 3 - This is all you should be concerend about... What is the last log
>> message you are getting from here? Then use xlog() to tell
>> you what you
>> need to know. If you are not even getting the first log
>> message then SER
>> is not aware that it is supposed to respond for whatever the
>> REGISTER has
>> requested. You will need to add it as an "alias" global
>> config option at
>> the top of the file.
>>
>> Then how do I let the ser to show the log message?
>> I modified the following to show the log in the terminal and I did add
>> "alias" for global config option at the top of ser.cfg.
>>
>> debug=3
>> fork=yes
>> log_stderror=yes
>>
>> alias="eplgroup.tara.ca"
>> alias="134.190.64.164"
>>
>> Thanks,
>> Yilan
>> > Couple of things to try....
>> >
>> >
>> > 1 - Check out xlog() so you can print out the URI, Contact, etc.
>> >
>> > 2 - Are you beheind NAT?
>> >
>> > 3 - This is all you should be concerend about... What is
>> the last log
>> > message you are getting from here? Then use xlog() to tell
>> you what you
>> > need to know. If you are not even getting the first log
>> message then SER
>> > is
>> > not aware that it is supposed to respond for whatever the
>> REGISTER has
>> > requested. You will need to add it as an "alias" global
>> config option at
>> > the top of the file.
>> >
>> >> if (uri==myself) {
>> >>
>> >> log(1,"in the served domain");
>> >> if (method=="REGISTER") {
>> >> log(1,"do the register work");
>> >>
>> >> #Uncomment this if you want to use digest authentication
>> >> if (!www_authorize("Ip address of
>> sip server",
>> >> "subscriber")) {
>> >> www_challenge("Ip address of
>> >> sip server",
>> >> "0");
>> >> log(1,"do www_challenge");
>> >> break;
>> >> };
>> >>
>> >> save("location");
>> >> log(1,"save in the location");
>> >> break;
>> >> };
>> >>
>> >> # native SIP destinations are handled using
>> >> our USRLOC DB
>> >> if (!lookup("location")) {
>> >> sl_send_reply("404", "Not Found");
>> >> log(1,"not found in the location");
>> >> break;
>> >> };
>> >> }else{
>> >> log(1,"not in the domain");
>> >> };
>> >
>> >
>> > ----------------------------------------
>> >
>> > Michael Shuler, C.E.O.
>> > BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
>> > 682 High Point Lane
>> > East Peoria, IL 61611
>> > Office: (217) 585-0357
>> > Cell: (309) 657-6365
>> > Fax: (309) 213-3500
>> > E-Mail: mike(a)bwsys.net
>> > Customer Service: (877) 976-0711
>> >
>> >
>> >
>> >
>> >
>> >
>> >> -----Original Message-----
>> >> From: serusers-bounces(a)lists.iptel.org
>> >> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of yilan(a)cs.dal.ca
>> >> Sent: Monday, September 27, 2004 7:19 PM
>> >> To: serusers(a)lists.iptel.org
>> >> Subject: [Serusers] Ser Register Problem
>> >>
>> >>
>> >> Dear All,
>> >>
>> >> I have the latest ser (0.8.14) on fedora 2 as the sip server
>> >> ,1 Mitel sip
>> >> phone and xpro soft phone as the sip phones. The problem is
>> >> that either
>> >> the mitel or xpro sip phones can't register in ser.
>> >>
>> >> I used "serctl ul show",nothing is registered.
>> >>
>> >> I used "sipsak -vv -s sip:6000@ip address of sipserver",it
>> >> shows "SIP/2.0
>> >> 404 Not Found"
>> >>
>> >> Then I added some log functions in ser.cfg where route
>> locates (listed
>> >> below).I found out when I used "sipsak -vv -s sip:6000@ip
>> address of
>> >> sipserver",the log functions execute first at uri==myself, then
>> >> (!lookup("location")) and I can see the logs in the terminal.
>> >>
>> >> But when I started the phone to communicate with the ser,
>> those logs
>> >> weren't show although I did see the register request from sip
>> >> phones to
>> >> ser from ngrep.
>> >>
>> >> So my first question is when the ser.cfg is executed. When I
>> >> used sipsak,
>> >> I could see ser.cfg is executed. When I started the sip
>> >> phone, I can't see
>> >> its execution.
>> >>
>> >> My second question is how to find out the reason for not
>> >> registering. I
>> >> used the IP address for the domain name. is it right to use?
>> >> Because I did
>> >> the test in local area network without dns.
>> >>
>> >> route{
>> >>
>> >> log(1,"Begin route\n");
>> >> if (!mf_process_maxfwd_header("10")) {
>> >> sl_send_reply("483","Too Many Hops");
>> >> log(1,"Too many hops\n");
>> >> break;
>> >> };
>> >> if ( msg:len > max_len ) {
>> >> sl_send_reply("513", "Message too big");
>> >> log(1,"Message too big\n");
>> >> break;
>> >> };
>> >>
>> >> record_route();
>> >> log(1,"record_route\n");
>> >> # loose-route processing
>> >> if (loose_route()) {
>> >> t_relay();
>> >> log(1,"loose route\n");
>> >> break;
>> >> };
>> >>
>> >> if (uri==myself) {
>> >>
>> >> log(1,"in the served domain");
>> >> if (method=="REGISTER") {
>> >> log(1,"do the register work");
>> >>
>> >> #Uncomment this if you want to use digest authentication
>> >> if (!www_authorize("Ip address of
>> sip server",
>> >> "subscriber")) {
>> >> www_challenge("Ip address of
>> >> sip server",
>> >> "0");
>> >> log(1,"do www_challenge");
>> >> break;
>> >> };
>> >>
>> >> save("location");
>> >> log(1,"save in the location");
>> >> break;
>> >> };
>> >>
>> >> # native SIP destinations are handled using
>> >> our USRLOC DB
>> >> if (!lookup("location")) {
>> >> sl_send_reply("404", "Not Found");
>> >> log(1,"not found in the location");
>> >> break;
>> >> };
>> >> }else{
>> >> log(1,"not in the domain");
>> >> };
>> >> # forward to current uri now; use stateful forwarding; that
>> >> # works reliably even if we forward from TCP to UDP
>> >> if (!t_relay()) {
>> >> sl_reply_error();
>> >> };
>> >>
>> >> }
>> >>
>> >> Thanks,
>> >> Yilan
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> Serusers mailing list
>> >> serusers(a)lists.iptel.org
>> >> http://lists.iptel.org/mailman/listinfo/serusers
>> >>
>> >
>> >
>>
>>
>
>
Hi,
Having tremendous trouble with SER + RTPProxy.
Running Fedora Core 1.
Have made RTPProxy using: make -f Makefile.gnu
Then run it fine by using: ./rtpproxy
Netstat shows:
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags Type State I-Node PID/Program name Path
unix 2 [ ACC ] STREAM LISTENING 11433 2443/rtpproxy /var/run/rtpproxy.sock
unix 43 [ ] DGRAM 1822 1947/syslogd /dev/log
unix 2 [ ] DGRAM 13503 2580/ser
unix 2 [ ] DGRAM 13500 2616/ser
unix 2 [ ] DGRAM 13497 2615/ser
...... [cropped]
When I start SER (version 0.8.14 - latest) I get this in the messages file:
Sep 28 16:03:46 testsip2 /sbin/ser[2582]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2582]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2582]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
Sep 28 16:03:46 testsip2 /sbin/ser[2583]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2583]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2583]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
Sep 28 16:03:46 testsip2 /sbin/ser[2584]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2584]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2584]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
Sep 28 16:03:46 testsip2 /sbin/ser[2585]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2585]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2585]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
Sep 28 16:03:46 testsip2 /sbin/ser[2586]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2589]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2587]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2588]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2589]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2590]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2591]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2592]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2586]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2587]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2593]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2594]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2595]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2596]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2588]: WARNING: rtpp_test: can't get version of the RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2597]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2598]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2599]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2600]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2589]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
Sep 28 16:03:46 testsip2 /sbin/ser[2601]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2602]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2603]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2604]: ERROR: send_rtpp_command: can't read reply from a RTP proxy
Sep 28 16:03:46 testsip2 /sbin/ser[2590]: WARNING: rtpp_test: can't get version of the RTP proxy
...... [cropped]
My ser.cfg if it helps is:
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
#alias=davidsimmons.co.uk
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "//lib/ser/modules/mysql.so"
loadmodule "//lib/ser/modules/nathelper.so"
loadmodule "//lib/ser/modules/sl.so"
loadmodule "//lib/ser/modules/tm.so"
loadmodule "//lib/ser/modules/rr.so"
loadmodule "//lib/ser/modules/maxfwd.so"
loadmodule "//lib/ser/modules/usrloc.so"
loadmodule "//lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "//lib/ser/modules/auth.so"
#loadmodule "//lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# inserted by klaus
if (method=="INVITE") {
record_route();
force_rtp_proxy();
/* set up reply processing */
t_on_reply("1");
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
#inserted by klaus
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
if (status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
Hope you can help as this is driving me mad now :(
Best Regards,
Dave
Hello.
I'm trying to compile the RTP proxy. I downloades the last version
from the CVS with :
[root@sipproxy root]# cvs
-d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co rtpproxy
cvs server: Updating rtpproxy
U rtpproxy/Makefile.am
U rtpproxy/Makefile.in
U rtpproxy/README
U rtpproxy/README.remote
U rtpproxy/aclocal.m4
U rtpproxy/config.h.in
U rtpproxy/configure
U rtpproxy/configure.ac
U rtpproxy/depcomp
U rtpproxy/install-sh
U rtpproxy/main.c
U rtpproxy/missing
U rtpproxy/mkinstalldirs
U rtpproxy/myqueue.h
[root@sipproxy root]#
Then i try to compile the program
[root@sipproxy rtpproxy]# make
make all-am
make[1]: Entering directory `/root/rtpproxy'
source='main.c' object='main.o' libtool=no \
depfile='.deps/main.Po' tmpdepfile='.deps/main.TPo' \
depmode=gcc /bin/sh ./depcomp \
gcc -DHAVE_CONFIG_H -I. -I. -I. -g -O2 -c `test -f 'main.c' || echo
'./'`main.c
main.c: In function `main':
main.c:1152: structure has no member named `ss_family'
make[1]: *** [main.o] Error 1
make[1]: Leaving directory `/root/rtpproxy'
make: *** [all] Error 2
[root@sipproxy rtpproxy]#
I'm using Red Hat Linux 7.3
Any ideas?
Ricardo Martinez.
I've just commited support for x86_64 (athlon64, athlon fx, opteron) for
both the stable (rel_0_8_14) and unstable (HEAD) cvs branches.
The x86_64 support is still experimental (it compiles, it starts, but I
haven't run any extensive tests).
Special thanks go to Michael Shuler for providing access to an Athlon64
machine.
Andrei
Hi Michael,
Thank you for your reply. I really appreciate it.
1 - Check out xlog() so you can print out the URI, Contact, etc.
I add xlog () in ser.cfg.Then for the sipsak command "sipsak -vv -s
sip:4500@eplgroup.tara.ca", it shows the following
5(6374) not found:time [Tue Sep 28 11:14:48 2004] method <OPTIONS> r-uri
<4500(a)eplgroup.tara.ca>
I guess sipsak doesn't send register method here.
But for the phone that is going to register for the ser, nothing shows
although I can see the phone sends register packet from ngrep.
2 - Are you beheind NAT?
NO.
3 - This is all you should be concerend about... What is the last log
message you are getting from here? Then use xlog() to tell you what you
need to know. If you are not even getting the first log message then SER
is not aware that it is supposed to respond for whatever the REGISTER has
requested. You will need to add it as an "alias" global config option at
the top of the file.
Then how do I let the ser to show the log message?
I modified the following to show the log in the terminal and I did add
"alias" for global config option at the top of ser.cfg.
debug=3
fork=yes
log_stderror=yes
alias="eplgroup.tara.ca"
alias="134.190.64.164"
Thanks,
Yilan
> Couple of things to try....
>
>
> 1 - Check out xlog() so you can print out the URI, Contact, etc.
>
> 2 - Are you beheind NAT?
>
> 3 - This is all you should be concerend about... What is the last log
> message you are getting from here? Then use xlog() to tell you what you
> need to know. If you are not even getting the first log message then SER
> is
> not aware that it is supposed to respond for whatever the REGISTER has
> requested. You will need to add it as an "alias" global config option at
> the top of the file.
>
>> if (uri==myself) {
>>
>> log(1,"in the served domain");
>> if (method=="REGISTER") {
>> log(1,"do the register work");
>>
>> #Uncomment this if you want to use digest authentication
>> if (!www_authorize("Ip address of sip server",
>> "subscriber")) {
>> www_challenge("Ip address of
>> sip server",
>> "0");
>> log(1,"do www_challenge");
>> break;
>> };
>>
>> save("location");
>> log(1,"save in the location");
>> break;
>> };
>>
>> # native SIP destinations are handled using
>> our USRLOC DB
>> if (!lookup("location")) {
>> sl_send_reply("404", "Not Found");
>> log(1,"not found in the location");
>> break;
>> };
>> }else{
>> log(1,"not in the domain");
>> };
>
>
> ----------------------------------------
>
> Michael Shuler, C.E.O.
> BitWise Communications, Inc. (CLEC) And BitWise Systems, Inc. (ISP)
> 682 High Point Lane
> East Peoria, IL 61611
> Office: (217) 585-0357
> Cell: (309) 657-6365
> Fax: (309) 213-3500
> E-Mail: mike(a)bwsys.net
> Customer Service: (877) 976-0711
>
>
>
>
>
>
>> -----Original Message-----
>> From: serusers-bounces(a)lists.iptel.org
>> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of yilan(a)cs.dal.ca
>> Sent: Monday, September 27, 2004 7:19 PM
>> To: serusers(a)lists.iptel.org
>> Subject: [Serusers] Ser Register Problem
>>
>>
>> Dear All,
>>
>> I have the latest ser (0.8.14) on fedora 2 as the sip server
>> ,1 Mitel sip
>> phone and xpro soft phone as the sip phones. The problem is
>> that either
>> the mitel or xpro sip phones can't register in ser.
>>
>> I used "serctl ul show",nothing is registered.
>>
>> I used "sipsak -vv -s sip:6000@ip address of sipserver",it
>> shows "SIP/2.0
>> 404 Not Found"
>>
>> Then I added some log functions in ser.cfg where route locates (listed
>> below).I found out when I used "sipsak -vv -s sip:6000@ip address of
>> sipserver",the log functions execute first at uri==myself, then
>> (!lookup("location")) and I can see the logs in the terminal.
>>
>> But when I started the phone to communicate with the ser, those logs
>> weren't show although I did see the register request from sip
>> phones to
>> ser from ngrep.
>>
>> So my first question is when the ser.cfg is executed. When I
>> used sipsak,
>> I could see ser.cfg is executed. When I started the sip
>> phone, I can't see
>> its execution.
>>
>> My second question is how to find out the reason for not
>> registering. I
>> used the IP address for the domain name. is it right to use?
>> Because I did
>> the test in local area network without dns.
>>
>> route{
>>
>> log(1,"Begin route\n");
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","Too Many Hops");
>> log(1,"Too many hops\n");
>> break;
>> };
>> if ( msg:len > max_len ) {
>> sl_send_reply("513", "Message too big");
>> log(1,"Message too big\n");
>> break;
>> };
>>
>> record_route();
>> log(1,"record_route\n");
>> # loose-route processing
>> if (loose_route()) {
>> t_relay();
>> log(1,"loose route\n");
>> break;
>> };
>>
>> if (uri==myself) {
>>
>> log(1,"in the served domain");
>> if (method=="REGISTER") {
>> log(1,"do the register work");
>>
>> #Uncomment this if you want to use digest authentication
>> if (!www_authorize("Ip address of sip server",
>> "subscriber")) {
>> www_challenge("Ip address of
>> sip server",
>> "0");
>> log(1,"do www_challenge");
>> break;
>> };
>>
>> save("location");
>> log(1,"save in the location");
>> break;
>> };
>>
>> # native SIP destinations are handled using
>> our USRLOC DB
>> if (!lookup("location")) {
>> sl_send_reply("404", "Not Found");
>> log(1,"not found in the location");
>> break;
>> };
>> }else{
>> log(1,"not in the domain");
>> };
>> # forward to current uri now; use stateful forwarding; that
>> # works reliably even if we forward from TCP to UDP
>> if (!t_relay()) {
>> sl_reply_error();
>> };
>>
>> }
>>
>> Thanks,
>> Yilan
>>
>>
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
The problem is that ser doesn't find any user in usrloc, because when
registering from my Cisco Phones the username in the usrloc database is
always "user@ip_adress_of_ser". I want "user@domain".
A problem of the Cisco 7905G Phones is, that they always use the
adress/ip of the sip proxy as domain.
Using my domain name as proxy adress in the phones is not possible
because the domain name has no dns entry
pointing at my SER Server.
>
>That has been removed for almost 10 months already. The domain part is
always used in registrar module.
>
>So long as you set use_domain=0 in the following modules, I don't
foresee any significant impact to your routing
>logic.
>
>- auth_db
>- auth_diameter
>- group
>- group_radius
>- uri_db
>- usrloc
>- vm (non CVS HEAD version)
>
>Regards,
>
>Zeus
>
>> -----Original Message-----
>> From: serusers-bounces(a)lists.iptel.org
>> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jankowski, Jan
>> Sent: Wednesday, 22 September 2004 9:34 PM
>> To: serusers(a)lists.iptel.org
>> Subject: [Serusers] No use_domain in cvs registrar?
>>
>>
>> Hi Serusers,
>>
>> there seems to be no "use_domain" option in the cvs version
>> of registrar anymore. I need ...."use_domain", 0) and ser
>> says "parameter <use_domain> not found in module
>> <registrar>". Am I too stupid or did you remove this feature? Why?
>>
>> With ser-0.8.14 everything was fine :-/
>>
>> Jan Jankowski
>>
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>>
