Hello! So glad to join your community. We've desided to use our server like a sipproxy server, but I have a priblem during installation of OpenSER. It looks like there are some troubles with Makefile. I have Pentium-D 3200 Suse server and whenever I try to compile OpenSER I have the following message:
linksysserver:/openser-1.0.1 # make all
Compiling action.c
gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=x86_64 -DNAME='"openser"' -DVERSION='"1.0.1"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 4.1.0"' -D__CPU_x86_64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c action.c -o action.o
action.c:1: error: bad value (x86_64) for -mtune= switch
make: *** [action.o] Error 1
Do you have any ideas on how to solve this problem?
Hello all. Over the last week, I’ve installed OpenSER on my Debian Sarge box with some degree of success. I’ve succeeded adding users to OpenSER, and I’ve stumbled my way through the openserctl commands to add my VOIP capable Ascend Max as a digital PSTN gateway. At this point I can only get PSTN calls to terminate from the PSTN to my un-NAT’d VOIP UA’s(grandstream HT-486’s). I’m still on my search to find the proper openserctl command to route all PSTN calls out my gateway. (any help here would be appreciated)
Also, could somebody point me in the right direction on where to find complete documentation on how to setup mediaproxy on OpenSER? I’ve followed the install instructions included with mediaproxy-1.7.2.tar.gz. but when I I attempt to start mediaproxy I get the following error….
sipproxy1:~# /etc/init.d/mediaproxy start
Starting SER MediaProxy server: mediaproxy/usr/bin/env: python: No such file or directory
/usr/bin/env: python: No such file or directory
My main goal is to get these HT486’s further out on the Internet and of course behind some kind of NAT device.
Thanks in advance for your help.
Marc
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.9.8/381 - Release Date: 7/3/2006
Hi Users,
I install the openser 1.0.1 and rtp proxy 0.3 in same Linux System
openser server is located with public id xx.xxx.xxx.xx of 192.168.2.2 ,
And UAC are outside the NAT,
When one UAC call to other UAC( are both in outside the NAT where openser
server), after the INVITE method get request by server, after 32 second its
hung up automatically, Voice is ok , and callee is hung upping, not caller,
UAC ( inside the nAT , openser server ) in not hung uping and voice is not
ok....
Where is the problem, in NAt with rtp or networking,
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# NAT detection
route(2);
if (!method=="REGISTER")
record_route();
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
append_hf("P-hint: outbound\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("xx.xxx.xxx.xxx", "subscriber")) {
www_challenge("xx.xxx.xxx.xxx", "0");
exit;
};
if (isflagset(5)) {
setflag(6);
# if you want OPTIONS natpings uncomment next
# setflag(7);
};
save("location");
exit;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')){
setflag(6);
};
if (isflagset(5)||isflagset(6)) {
route(3);
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]{
force_rport();
if (nat_uac_test("19")) {
if (method=="REGISTER") {
fix_nated_register();
} else {
fix_nated_contact();
};
setflag(5);
};
}
route[3] {
if (is_method("BYE|CANCEL")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
force_rtp_proxy();
t_on_failure("1");
};
if (isflagset(5))
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
t_on_reply("1");
}
failure_route[1] {
if (isflagset(6) || isflagset(5)) {
unforce_rtp_proxy();
}
}
onreply_route[1] {
if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") {
force_rtp_proxy();
}
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
if (isflagset(6)) {
fix_nated_contact();
}
exit;
}
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.
www.hyperion-tech.com
+91-9985077535
Hello,
First, thanks to the developer of the openser software for their
work. I will appreciate it when my configuration will be working ;)
I would like to configure sip on my server but I'm a bit lost.
I've installed openser on it but I cannot have it to do what
I want to do.
So this is what I want to do:
- I own a virtual server which host my domain cosinux.org . I would
like to create myself a sip address dam(a)cosinux.org.
- I would like to be able to send/receive sip call at that address
from work. My work network is firewalled, however, I have a vtun
tunnel with my server.
- I would like to be able to send/receive sip call at that address
from home. I have dsl at home and I can configure nat as I want to
(port forwarding for example).
>From what I understand, openser is used as registrar. It means
my sip client connects to it, tells him that dam(a)cosinux.org is online
and openser then redirects call to my client. This will work from
my home network if I configure nat correctly.
For my work network, I need a sip proxy so that each packet is routed
by my server to my client, because others sip user agent cannot see
my IP address.
If I am right, what I have to do is
- install openser on my server and tell him to handle cosinux.org
sip address.
- install mediaproxy to relay the sip connection from my work.
So, does this seem sensible ? Am I doing something wrong or missing
something ?
Please CC me as I'm not suscribed to the list.
Thank you very much,
dam
--
Damien MERENNE <dam(a)cosinux.org>
http://www.cosinux.org/blogs/dam/
CNN was originally created as the "Chuck Norris Network" to update Americans
with on-the-spot ass kicking in real-time.
Hi SER users
iam using ser-0.9.6 when i tried with x-lite , grandstream phones , i can
make calls and recieve calls
now iam using one new phone it is 'ip-300' SIP phone i can register taht
phone and i can make calls to that 'ip-300' but when I tried to make calls
from that phone it is not success
my ngrep messages are as follows :
U 81.21.34.34:46117 -> 81.21.33.35:5060
INVITE sip:32331003@81.21.33.35 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9.
Max-Forwards: 70.
User-Agent: IP-300.
From: "32331001" <sip:32331001@81.21.33.35>;tag=ib0b3FffOYuu1C8n.
To: "32331003" <sip:32331003@81.21.33.35>.
Call-ID: HTqDhmIJuXyigmii(a)192.168.0.74.
Contact: <sip:32331001@192.168.0.74:5060>.
CSeq: 1 INVITE.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 289.
.
v=0.
o=- 32362173 86500032 IN IP4 192.168.0.74.
s=SIP CALL.
c=IN IP4 192.168.0.74.
t=0 0.
m=audio 8000 RTP/AVP 18 4 0 8 3 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 81.21.33.35:5060 -> 81.21.34.34:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bKH7w0kkrIoD6CyPP9;received=
81.21.34.34.
From: "32331001" <sip:32331001@81.21.33.35>;tag=ib0b3FffOYuu1C8n.
To: "32331003" <sip:32331003@81.21.33.35>;tag=
74961b5b71b6ddce908b9155b956083f.a2b9.
Call-ID: HTqDhmIJuXyigmii(a)192.168.0.74.
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="81.21.33.35",
nonce="44ae1171fed2ef64959f27604e81bf11661bd0b5".
Content-Length: 0.
Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=19217 req_src_ip=
81.21.34.34 req_src_port=46117 in_uri=sip:32331003@81.21.33.35 out_uri=
sip:32331003@81.21.33.35 via_cnt==1".
.
#
U 81.21.33.35:5060 -> 81.21.34.34:5060
SIP/2.0 410 Gone.
Via: SIP/2.0/UDP 192.168.0.74:5060;received=81.21.34.34
;branch=z9hG4bKZRDQYKLawgmypijq.
From: <sip:32331001@81.21.33.35>;tag=0eZhJPf62anG4L1n.
To: ravi <sip:32331003@81.21.33.35:5062>;tag=3806266128.
Contact: <sip:32331003@81.21.34.34:5062>.
Record-Route: <sip:32331001@81.21.33.35:5060;nat=yes;ftag=3806266128;lr=on>.
Call-ID: 117AD09F-713F-41BF-B61F-7AFFA70FD4C5(a)192.168.0.79.
CSeq: 2 INVITE.
Server: X-Lite release 1105x.
Content-Length: 0.
.
What does this mean and "this phone works with Asterisk"
I think i made mistake some where any body any clues please
Thank You.
Regards
crap, forgot to reply to all again :(
-------- Original Message --------
Subject: Re: [Serusers] Can anyone explain this error?
Date: Thu, 06 Jul 2006 15:42:38 +0200
From: nick <nick(a)mobilia.it>
To: sip <sip(a)arcdiv.com>
References: <7EB6E3B0D22DF54585D1CEB9642E3198D85D13(a)owa.koc.net>
<44ACF1C3.1080301(a)mobilia.it> <20060706110123.M63766(a)infinideas.com>
sip wrote:
> Just to make sure, try:
>
> mysql -u ser -h localhost -p
>
> And see if you still get access denied. Connecting directly via the socket and
> connecting as though to an external host (in this case, localhost) actually
> give you different results. If that works, we can ensure that's not part of
> the equation.
>
> N.
>
>
Yes, with the password I was able connect..
[root@sipserver ~]# mysql -u ser -h localhost -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 6 to server version: 4.1.12
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
mysql>
hello once more...
since the record-route-header contains a maddr=<ip_of_server>, shouldn't OpenSER
use that address directly?
from rfc3261, section 4.
--
The procedures here are invoked when a client needs to send a request
to a resource identified by a SIP or SIPS (secure SIP) URI. This URI
can identify the desired resource to which the request is targeted
(in which case, the URI is found in the Request-URI), or it can
identify an intermediate hop towards that resource (in which case,
the URI is found in the Route header).
...
We define TARGET as the value of the maddr parameter of the URI, if
present, otherwise, the host value of the hostport component of the
URI. It identifies the domain to be contacted.
--
or am i doing something wrong here?
best regards,
/Staffan
> -----Ursprungligt meddelande-----
> Från: users-bounces(a)openser.org
> [mailto:users-bounces@openser.org] För Norman Brandinger
> Skickat: den 6 juli 2006 14:57
> Till: users(a)openser.org
> Ämne: Re: [Users] Route-header DNS lookup?
>
> Hi Kerker,
>
> Perhaps there needs to be SRV records in the DNS server that
> manages iptel1.ipatl.se.
>
> Below are example SRV records. If this is your first time
> working with SRV records, I strongly suggest reading some of
> the online doc that deals with them.
>
> _sip._tcp.iptel1.ipatl.se. IN SRV 0 0 5060
> sip.iptel1.ipatl.se.
> _sip._udp.iptel1.ipatl.se. IN SRV 0 0 5060
> sip.iptel1.ipatl.se.
>
> Regards,
> Norm
>
> Kerker Staffan wrote:
> > hi
> > i recently bounced into this problem, and i'm not sure here.
> > i'm running the openser-devel, with the cacheless
> db_mode=3. (works
> > fine btw)
> >
> > the record-route header received by the proxy on the other side
> > (SNOM4S), inserts the domain name (iptel1.ipatl.se) and not the
> > hostname (sip.iptel1.ipatl.se) in the record-route header,
> and uses the maddr=<ip_of_server> with the actual server IP address.
> >
> > now, when my client (behind the OpenSER) replies with an
> ACK to the
> > incomming OK, it uses the Route-header recieved in the
> RR-header, and
> > sends the ACK to OpenSER. i then get the following errors
> in OpenSER.
> >
> > ---
> > /usr/local/sbin/openser[3583]: ERROR: mk_proxy: could not
> resolve hostname: "iptel1.ipatl.se"
> > /usr/local/sbin/openser[3583]: ERROR: uri2proxy: bad host
> name in URI
> >
> <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060;ma
> ddr=172.2
> > 8.248.66;transport=udp;lr>
> > ---
> >
> > the ACK i sent look like this:
> >
> > ---
> > Request-Line: ACK sip:2307@iptel1.ipatl.se;gruu=6fg9n6dl SIP/2.0
> > Via: SIP/2.0/UDP
> 172.28.248.52:2051;branch=z9hG4bK-d96b1fvapkyn;rport
> > Route: <sip:172.28.248.10;lr=on;ftag=li9buf1i4p>
> > Route:
> <sip:4ffec4ce755c218a72228c6643cb3b6b@iptel1.ipatl.se:5060
> ;maddr= 172.28.248.66;transport=udp;lr>
> > From: "Snom 2652" <sip:2652@ipatl.se>;tag=li9buf1i4p
> > To: <sip:2307@ipatl.se>;tag=hvseiz7kgb
> > Call-ID: 3c269d83900b-xj3ild14y880@snom360
> > CSeq: 1 ACK
> > Max-Forwards: 70
> > Contact: <sip:2652@172.28.248.52:2051;line=cp4a7ljd>
> > Content-Length: 0
> > ---
> >
> > as far as i understand, according the rfc 3263, the
> route-header may
> > contain domain name that has to be resolved using SRV.
> >
> > ---
> > "6 Constructing SIP URIs
> >
> > In many cases, an element needs to construct a SIP URI
> for inclusion
> > in a Contact header in a REGISTER, or in a Record-Route
> header in an
> > INVITE. According to RFC 3261 [1], these URIs have to have the
> > property that they resolve to the specific element that inserted
> > them. However, if they are constructed with just an IP
> address, for
> > example:
> >
> > sip:1.2.3.4
> >
> > then should the element fail, there is no way to route
> the request or
> > response through a backup.
> >
> > SRV provides a way to fix this. Instead of using an IP
> address, a
> > domain name that resolves to an SRV record can be used:
> >
> > sip:server23.provider.com"
> > ---
> >
> > now, OpenSER only asks DNS for an A record of the name
> recieved in the
> > route header, and since that's a domain name, it's
> unresolvable, and so the ACK is never sent.
> >
> > any hints or clues?
> >
> > best regards,
> > /Staffan Kerker
> >
> >
> > --
> > Staffan Kerker
> > Saab Communications, Växjö
> > p. +46 470 42185
> > c. +46 705 391365
> > m. staffan.kerker(a)saabgroup.com
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users(a)openser.org
> > http://openser.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> >
>
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
Hi all!
I'm sending this in my message:
Authorization: Digest
username="test2",realm="lab.pt",nonce="",uri="sip:lab.pt",response=""
Then,I tried to get the value of the select
"@hf_value.authorization.username" and all I've got was a empty string.
Then, for testing, I printed "@hf_value.authorization" and only got this:
-->> Digest username="test2"
Is this working fine? tks in advance
Regard,
Luis Silva