Hello,
I would like to know companies that have worked with the iptel.org
applications along with individuals interested in building out a complete
unified communications network.
This is for pay and also possible hiring on of developers and system
administrators.
We are reviewing building out with both the iptel and Vovida platforms, and
also have our own cisco gateways, gatekeepers, and unified communications
platforms.
We need billing, provisioning and individuals who can continue in the
development of the platforms.
Please e-mail me at HYPERLINK "mailto:dfeuer@cox.net"dfeuer(a)cox.net
Sincerely,
Don Feuer
CentreOne
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Hi,
As i mentioned earlier that stooping ser failed but starting it works ok,
i checked the syslog error and this is what i see:
The syslog error indicates:
module version mismatch for /usr/lib/ser/modules/dbtext.so;
core:0.8.11;module:0.8.12
I copied the dbtext.so file from the ser-0.8.12 folder into the
/usr/lib/ser/modules folder but this is not of any help.How do i fix this?
I have the ser-0.8.12-0.i386.rpm and ser-0.8.12_src.tar.gz. Do i need to
download the entire source tree of ser from the cvs login.I tried the
anonymous connection but it keeps timing out.
Please help.
Thanks,
Annie
Hi,
I have this network configuration
SIP Phone (ph.no: 3322) -> SER -> SIPH323 -> router -> Cisco CM ->Cisco Skinny phones(ph.no: 1133).
Their corresponding IP address are
192.168.6.103 -> 192.168.6.100 (port:5060) -> 192.168.6.100 (port 5080 for sip side of SIPH323 converter) -> etc
The problem arises in SER -> SIPH323 communication. SIPH323 doesnot get any SIP recognisable packet that SER sends.
In the SER configuration, I check for uri sip:1133@192.168.6.100 and then I forward to port 5080 where SIPH323 listens for SIP side messages.
I have attached the following:
1) My ser configuration 2)ngrep output on lo (loopback interface) 3) ngrep output on eth0.
I would appreciate if you could tell me what is happening and how to solve this problem. I also would like to know if my SER configuartion is correct. I am sure SIPH323 is configured properly.
If you see the ngrep on loopback interface, I could see a message going from 192.168.6.100 -> 192.168.6.100 3:3. This message has some weird characters before the INVITE message. I guess SIPH323 gets this message and doenot respond to it because it is not a valid SIP Invite message. I would like to know why this happens.
ATTACHED FILES:
1) SER.cfg
.
.
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 ( len_gt(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 (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;
};
if(uri=~"^sip:11[0-9]*@192.168.6.100")
{
forward(192.168.6.100, 5080);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# 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();
};
}
};
# 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)
2) ngrep on lo (loopback interface):
interface: lo (127.0.0.0/255.0.0.0)
#
U 192.168.6.100:5060 -> 192.168.6.100:5080
INVITE sip:1133@192.168.6.100 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133@192.168.6.100;ftag=925f0b001c06323a74
788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=
0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100>;tag=925f0b001c06323a74788-57c5
9f3f..To: <sip:1133@192.168.6.100>..Call-ID: 000b5f92-63c00
003-69765ab4-289e7e69@192.168.6.103..Date: Tue, 20 Jan 2004
23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent:
Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip
:3322@192.168.6.103:5060..Content-Type: application/sdp..Co
ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge
nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19
2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp
map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:
101 0-11..
#
I 192.168.6.100 -> 192.168.6.100 3:3
....E..`..@.@..d...d...d.....L.7INVITE sip:1133@192.168.6.1
00 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:1133@192.1
68.6.100;ftag=925f0b001c06323a74788-57c59f3f;lr=on>..Via: S
IP/2.0/UDP 192.168.6.100;branch=0..Via: SIP/2.0/UDP 192.168
.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133@192.168
.6.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.1
68.6.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 IN
VITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accep
t: application/sdp..C
#
U 192.168.6.100:5060 -> 192.168.6.100:5080
INVITE sip:1133@192.168.6.100 SIP/2.0..Max-Forwards: 10..Re
cord-Route: <sip:1133@192.168.6.100;ftag=925f0b001c06323a74
788-57c59f3f;lr=on>..Via: SIP/2.0/UDP 192.168.6.100;branch=
0..Via: SIP/2.0/UDP 192.168.6.103:5060..From: "User ID Bala
ji" <sip:3322@192.168.6.100>;tag=925f0b001c06323a74788-57c5
9f3f..To: <sip:1133@192.168.6.100>..Call-ID: 000b5f92-63c00
003-69765ab4-289e7e69@192.168.6.103..Date: Tue, 20 Jan 2004
23:34:41 GMT..CSeq: 101 INVITE..Expires: 180..User-Agent:
Cisco-SIP-IP-Phone/2..Accept: application/sdp..Contact: sip
:3322@192.168.6.103:5060..Content-Type: application/sdp..Co
ntent-Length: 225....v=0..o=CiscoSystemsSIP-IPPhone-UserAge
nt 11416 8413 IN IP4 192.168.6.103..s=SIP Call..c=IN IP4 19
2.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0 8 18 101..a=rtp
map:0 pcmu/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:
101 0-11..
...It repeats the same messages
3) ngrep on eth0:
interface: eth0 (192.168.6.0/255.255.255.0)
###############
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133@192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
#
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133@192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
#
U 192.168.6.103:49834 -> 192.168.6.100:5060
INVITE sip:1133@192.168.6.100 SIP/2.0..Via: SIP/2.0/UDP 192.1
68.6.103:5060..From: "User ID Balaji" <sip:3322@192.168.6.100
>;tag=925f0b001c06323a74788-57c59f3f..To: <sip:1133@192.168.6
.100>..Call-ID: 000b5f92-63c00003-69765ab4-289e7e69(a)192.168.6
.103..Date: Tue, 20 Jan 2004 23:34:41 GMT..CSeq: 101 INVITE..
Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: appli
cation/sdp..Contact: sip:3322@192.168.6.103:5060..Content-Typ
e: application/sdp..Content-Length: 225....v=0..o=CiscoSystem
sSIP-IPPhone-UserAgent 11416 8413 IN IP4 192.168.6.103..s=SIP
Call..c=IN IP4 192.168.6.103..t=0 0..m=audio 24234 RTP/AVP 0
8 18 101..a=rtpmap:0 pcmu/8000..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-11..
###
...it repeats these messages
Thanks,
Balaji
The syslog error indicates:
module version mismatch for /usr/lib/ser/modules/dbtext.so;
core:0.8.11;module:0.8.12
How do i fix this?
I have the ser-0.8.12-0.i386.rpm and ser-0.8.12_src.tar.gz. Do i need to
download the entire source tree of ser from the cvs login.I tried the
anonymous connection but it keeps timing out.
Please help.
Thanks,
Annie
Balaji,
Use Ethereal at http://www.ethereal.com/ to capture the traces.
Then it gives you the option to save the dump in a libpcap format.
Use that dump file as the input to your sip scenario generator.
Hope this helps.
Darshan.A.Uka
Multimedia Labs
DePaul University
Chicago,IL 60604
----Original Message Follows----
From: Jiri Kuthan <jiri(a)iptel.org>
To: bbthog2(a)uky.edu, serusers(a)lists.iptel.org
Subject: Re: [Serusers] Reg. SIP Scenario Generator
Date: Tue, 20 Jan 2004 22:26:45 +0100
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At 11:55 PM 1/19/2004, Balaji Bapulal Thoguluva wrote:
>Hi,
>
> I would like to know if anybody has used SIP Scenario Generator
software. I saw this from this website http://www.iptel.org/~sipsc/
I did, I liked it.
> If any body has already used it, I would like to know what kind of
file is given as input to it. The document says any libpcap formatted file
can be given as input.
yes, it is what the doc tells.
>I guessed tcpdump files produces lipbcap formatted files. But it doesnt
seem to work.
it does.
>I would appreciate if anybody could tell about what is lipcap formatted
file and how I can generate it.
use tcpdump.
-jiri
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_________________________________________________________________
Let the new MSN Premium Internet Software make the most of your high-speed
experience. http://join.msn.com/?pgmarket=en-us&page=byoa/prem&ST=1
I've used it in windows and it works pretty good but it would be better
if it had a gui. All I did was save an ethereal file and fed it to
sip_scenario.exe at the command line and it genereated an html SIP
step-ladder trace.
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Tuesday, January 20, 2004 2:27 PM
To: bbthog2(a)uky.edu; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Reg. SIP Scenario Generator
At 11:55 PM 1/19/2004, Balaji Bapulal Thoguluva wrote:
>Hi,
>
> I would like to know if anybody has used SIP Scenario Generator
> software. I saw this from this website http://www.iptel.org/~sipsc/
I did, I liked it.
> If any body has already used it, I would like to know what kind of
file is given as input to it. The document says any libpcap formatted
file can be given as input.
yes, it is what the doc tells.
>I guessed tcpdump files produces lipbcap formatted files. But it doesnt
seem to work.
it does.
>I would appreciate if anybody could tell about what is lipcap
formatted file and how I can generate it.
use tcpdump.
-jiri
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Mike,
I did as u suggested, still when i do a restart, Stopping ser: Failed but
Starting ser: OK.
Any tips on where i should look to solve the problem?
Thanks,
Annie
Hi,
I would like to know if anybody has used SIP Scenario Generator software. I saw this from this website http://www.iptel.org/~sipsc/
If any body has already used it, I would like to know what kind of file is given as input to it. The document says any libpcap formatted file can be given as input. I guessed tcpdump files produces lipbcap formatted files. But it doesnt seem to work. I would appreciate if anybody could tell about what is lipcap formatted file and how I can generate it.
Thanks,
Balaji
hi, first post here after reading thru the archives and whatnot.
im a ser-n00b, so go easy. i installed from source with the -all option for
all modules. everything is where it should be, i think.
basically, i am getting MySQL errors when trying to access serweb.
--
Warning: Access denied for user: 'ser@Lima' (Using password: YES) in
/var/www/html/phplib/db_mysql.inc on line 73
Database error: pconnect(Lima, ser, $Password) failed.
MySQL Error: ()
Session halted.
--
the ser db is created, along with a 'ser' user and the default heslo passwd.
i used phpMyAdmin to grant ser full permissions over the ser db. i suspected
the problem is the appending of @Lima to the user name was problematic, so i
created a 'ser@Lima' user and gave it permissions for the ser db as well.
still no luck.
here are some relevant bits from ser.cfg. if you need me to post logs, other
config files, let me know. thanks for reading!
----
$this->db_host="Lima"; //database host
$this->db_name="ser"; //database name
$this->db_user="ser"; //database conection user
$this->db_pass="heslo"; //database conection password
$this->root_path="/serweb/";
/* root uri of your server */
/* $this->root_uri="http://Lima/"; */
$this->root_uri="http://10.0.0.54/";
/* where is your zone file on your server ? */
$this->zonetab_file = "/usr/share/zoneinfo/zone.tab";
/* serweb will send confirmation emails and SIP IMs -- what
sender
address should it claim ?
should appear in them ?
*/
$this->mail_header_from="registrar(a)mydomain.org";
$this->web_contact="sip:daemon@Lima";
/* spool directory with voicemail messages */
$this->voice_silo_dir = '/var/spool/voicemail/';
/* directory with voicemail greetings */
$this->greetings_spool_dir = '/var/greetings/';
/* serweb talks to SER via FIFO -- this is FIFO's name,
it must have the same value as SER's fifo config param
*/
$this->fifo_server="/tmp/ser_fifo";
----