I understand that by default ser doesn't write much log output and what
is does ends up in /var/log/messages via the syslog facility.
I also understand that one can get detailed logs printed to stderr by
changing some settings in ser.cfg and restarting ser.
Does anyone have a suggestion as to the best way to get detailed logs
dumped to a file so that I can tail them etc?
most likely caused by a misrouted ACK. -jiri
At 10:36 AM 3/12/2004, Manuel Goertz wrote:
>Hi all,
>
>I have a problem calling from a sipset to a ISDN subscriber over
>a CISCO 1760 GW.
>The following setup is used.
>UA ---> GW ---> ISDN
>The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface
>and a standard ISDN subscriber.
>The UA is registered with the SER server.
>All numeric userparts of the SIP URI are rewritten and routed to the GW.
>The GW's BRI interface is connected to the PSTN.
>The call signaling seems to work as the SIP phone indicates ringing
>and the ISDN phone is ringing. After picking up the hook of the ISDN
>phone the UA shows "In Call". But after a second the call is
>terminated. The log shows that the GW sends to both side the call
>termination messages. TX -> DISCONNECT "Normal Call Clearing" to the
>ISDN side and a BYE message to the SIP side.
>The signaling in short:
>
>UA GW ISDN
>INVITE -> |
> <- 100 Try
> | TX -> SETUP
> | RX <- CALL_PROC
> | RX <- ALERTING
> <- 183 Sess |
> | RX <- CONNECT
> | TX -> CONNECT_ACK
> <- 200 OK |
> Milliseconds later !
> | TX -> DISCONNECT
> | RX <- RELEASE
> | TX -> RELEASE_COMP
> <- BYE |
>200 OK -> |
>
>
>Any hints how to solve this problem.
>
>Thanks
>
> Manuel
>
>
>
>
>
>
>--
>+KOM----------------------------------------------------------------+
>|Manuel Görtz Merckstrasse 25|
>|Darmstadt University of Technology 64283 Darmstadt, Germany|
>|Multimedia Communications Tel: (+49) 6151 16-5175|
>|Multimedia Networking & Distribution Fax: (+49) 6151 16-6152|
>+----------------------------------------------------------------KOM+
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
I just started with SER. I downloaded and compiled
SER with MYSQL auth. Everything worked fine. I then
compiled and installed the radius server and made sure
it works. Then I compiled SER with Radius options,
and then changed the SER configuration file based on
the howto file. But when I run sectl start I get the
following error:
#serctl start
Starting SER : cat: /var/run/ser.pid: No such file or
directory
started pid()
and ser is not running. If I comment out everything
related to Radius, everything runs.
I probably have a bad configuration. Can someone
please post a copy of a working ser configuration file
with Radius authentication, authorization, and
accounting and a couple of outside gateway routing
example all enabled?
__________________________________
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I'm using the script from http://lists.iptel.org/pipermail/serusers/2004-February/005996.html
(the first ser.cfg). I'm trying to figure out where to include my SIP outgoing calls to my Asterisk PSTN server. For example, let's say I want to transfer a SIP call for a local XXX-XXXX number to pstn.xyz.com port 5050. How would I set that up?
Additionally, I'm rather confused as to which section I would put a dialplan into for say connecting to FWD users, etc. Again, like before if I wanted to send a certain call to a certain termination point, let's say 407-xxx-xxxx, I want to send that to one IP, and 954-xxx-xxxx I want to sent to a different one. Where are those statements supposed to be?
Thanks a lot...
Hi all,
I have a problem calling from a sipset to a ISDN subscriber over
a CISCO 1760 GW.
The following setup is used.
UA ---> GW ---> ISDN
The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface
and a standard ISDN subscriber.
The UA is registered with the SER server.
All numeric userparts of the SIP URI are rewritten and routed to the GW.
The GW's BRI interface is connected to the PSTN.
The call signaling seems to work as the SIP phone indicates ringing
and the ISDN phone is ringing. After picking up the hook of the ISDN
phone the UA shows "In Call". But after a second the call is
terminated. The log shows that the GW sends to both side the call
termination messages. TX -> DISCONNECT "Normal Call Clearing" to the
ISDN side and a BYE message to the SIP side.
The signaling in short:
UA GW ISDN
INVITE -> |
<- 100 Try
| TX -> SETUP
| RX <- CALL_PROC
| RX <- ALERTING
<- 183 Sess |
| RX <- CONNECT
| TX -> CONNECT_ACK
<- 200 OK |
Milliseconds later !
| TX -> DISCONNECT
| RX <- RELEASE
| TX -> RELEASE_COMP
<- BYE |
200 OK -> |
Any hints how to solve this problem.
Thanks
Manuel
--
+KOM----------------------------------------------------------------+
|Manuel Görtz Merckstrasse 25|
|Darmstadt University of Technology 64283 Darmstadt, Germany|
|Multimedia Communications Tel: (+49) 6151 16-5175|
|Multimedia Networking & Distribution Fax: (+49) 6151 16-6152|
+----------------------------------------------------------------KOM+
Hi,
We've got an intertex IX66 SIP User agent that we are trying to register on
SER (0.8.12 CVS). I've attached the ngrep output of the registeration
attempt, it reports "Noisy feedback tells". However it works fine with other
clients. (We're using radius based authentication)
I've tried adding xlog commands to ser.cfg, though nothing appears in the
log, though ngrep picks it up...
What could the problem be ??
Regards,
Alan
#
U 82.32.34.228:5060 -> 1.2.3.81:5060
REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran
ch=z9hG4bK0fc1a0e27d3f673ab0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdat
a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal
l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49
b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 101
REGISTER..User-Agent: CSCO/4..Contact:
<sip:kTHowCXnSy7o@82.32.34.228>;expi
res=3600..Content-Length: 0..Expires: 3600....
#
U 1.2.3.81:5060 -> 82.32.34.228:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
82.32.34.228:5060;branch=z9hG4bK
0fc1a0e27d3f673ab0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdata=Qv4u+bUN
yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T
o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.0b47..
Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 101
REGIS
TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e6d0eeb5ff
c4083b4ba14c7d03479746aeb", qop="auth"..Server: Sip EXpress router (0.8.12
(i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe
edback tells: pid=22432 req_src_ip=82.32.34.228 req_src_port=5060
in_uri=s
ip:sip.comout_uri=sip:sip.com via_cnt==2"....
#
U 82.32.34.228:5060 -> 1.2.3.81:5060
REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran
ch=z9hG4bK60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdat
a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal
l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49
b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 102
REGISTER..User-Agent: CSCO/4..Contact:
<sip:kTHowCXnSy7o@82.32.34.228>;expi
res=3600..Authorization: Digest username="12349000",realm="sip.com"
,uri="sip:sip.com",response="a597e49eb1578fad3ed27ec38beb460c",nonce=
"40509e6d0eeb5ffc4083b4ba14c7d03479746aeb",cnonce="55a1ffcb",qop=auth,nc=00
000001,algorithm=md5..Content-Length: 0..Expires: 3600....
#
U 82.32.34.228:5060 -> 1.2.3.81:5060
REGISTER sip:sip.com SIP/2.0..Via: SIP/2.0/UDP 82.32.34.228:5060;bran
ch=z9hG4bK60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdat
a=Qv4u+bUNyCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sipcal
l.co.uk..To: sip:12349000@sip.com..Call-ID: 000ab761-6341000e-3ea49
b31-20169200@192.168.1.105..Date: Thu, 11 Mar 2004 17:09:21 GMT..CSeq: 102
REGISTER..User-Agent: CSCO/4..Contact:
<sip:kTHowCXnSy7o@82.32.34.228>;expi
res=3600..Authorization: Digest username="12349000",realm="sip.com"
,uri="sip:sip.com",response="a597e49eb1578fad3ed27ec38beb460c",nonce=
"40509e6d0eeb5ffc4083b4ba14c7d03479746aeb",cnonce="55a1ffcb",qop=auth,nc=00
000001,algorithm=md5..Content-Length: 0..Expires: 3600....
#
U 1.2.3.81:5060 -> 82.32.34.228:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
82.32.34.228:5060;branch=z9hG4bK
60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdata=Qv4u+bUN
yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T
o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.ecd2..
Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 102
REGIS
TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e6f4a04de1
ce336dd980cc5962ca10bb63b", qop="auth"..Server: Sip EXpress router (0.8.12
(i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe
edback tells: pid=22433 req_src_ip=82.32.34.228 req_src_port=5060
in_uri=s
ip:sip.comout_uri=sip:sip.com via_cnt==2"....
#
U 1.2.3.81:5060 -> 82.32.34.228:5060
SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
82.32.34.228:5060;branch=z9hG4bK
60a2f914cc748de2b0e7b18b82c39281.0..Via: SIP/2.0/UDP
0.0.0.0;xdata=Qv4u+bUN
yCJEXOPOiyVkU6z-szKpIpeClHN0fsq68N4_..From: sip:12349000@sip.com..T
o: sip:12349000@sip.com;tag=cc34cfa6d75af81b0409220a2ba8e06a.ecd2..
Call-ID: 000ab761-6341000e-3ea49b31-20169200@192.168.1.105..CSeq: 102
REGIS
TER..WWW-Authenticate: Digest realm="sip.com", nonce="40509e711121e42
910dfd8d6d4ee741bf6615d18", qop="auth"..Server: Sip EXpress router (0.8.12
(i386/linux))..Content-Length: 0..Warning: 392 1.2.3.81:5060 "Noisy fe
edback tells: pid=22434 req_src_ip=82.32.34.228 req_src_port=5060
in_uri=s
ip:sip.comout_uri=sip:sip.com via_cnt==2"....
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Hi,
I'm using a WLAN sip phone, and I'm connecting to my SIP server from
different locations.
In some locations, when I connect, place a call, then hang up (from the
sip phone) ser receives my BYE message but send me a 407 proxy
authentication required message, this message is not answered by my sip
phone.
>From other locations, the BYE message is received by ser and relayed to
he other end point, without problems.
In which case a proxy authentication is required after a BYE ?
Thanks.
Samy.
Hi,
I am struggling to get SIP (session initiation protocol) VoIP software
(SIP Express Router + XTen/KPhone/Linphone) and hardware (Grandstream
SIP IP Phone) to work with Shorewall. The best result I have achieved
is when I can call (or someone call me), hear sound, but my talk is not
transmitted for weird reason.
I have quite standard configuration - two-card Linux PC with shorewall
(one static external IP + static local IPs 192.168.0.xxx), opened ports
are 5060 tcp/udp, 5004 udp (for Grandstream), and 8000-8020 udp (for
XTen, Linphone).
SER itself is hosted by the independent provider, and other people who
have another router/firewall software do not have similar problem.
It looks like something is wrong with NAT or masquarading. May be
someone will be kind to e-mail (or post on this mailing list) their
Shorewall configs (masq, nat, rules, etc.)? My e-mail is
andreil1(a)starlett.lv
Thanks in advance.
*********************************************
* Best Regards --- Andrei Verovski
*
* Personal Home Page
* http://snow.prohosting.com/guru4mac/
* Mac, Linux, DTP, Development, IT WEB Site
*********************************************
Hi,
I ve recently started with SER. So far I ve managed to
install MYSQL and SER on REDHAT LINUX ver 9. I can also
start the MYSQL Server and SER. The problem I am facing
now is that when ever i go to /usr/sbin and give the
command ./ser_mysql.sh <variable> to create the mysql
table it gives me error messageas follows:
this is the message with ./ser_mysql.sh create>>
> creating database ser ...
> ./ser_mysql.sh: line 80: mysql: command not found.
With"reinstall" gives me different error message which is
as follows:
> ./ser_mysql.sh: line 102: mysqldump: command not found
> ser backup dump failed.
Can anyone help me on this I know i am missing
something but I am not quite sure.And can any one please
tell me how to configure Messenger service as well.
Any help on this I'll be really greatful.
Thanks
shahzad
Before I reinvent the wheel and go write it, is there a utility which will
simulate making & receiving calls (i.e. SIP conversations, **plus RTP**) in
scale (to answer a question like "will my SIP+RTPproxy deployment scale for
{100,1000,10000,100000} users using the following parameters for average
call duration, % phones in use etc."
If not, anyone have any recommendations for a (free) stack that does a
decent UAC/UAS job, including audio, and will run without GUI?
Alex
