Hello,
I have notice that when I configure http-digest authentication for INVITE messages in OpenSER 1.2.2, there is a big difference in the call throughput when you use a database in a remote machine. For example, when using a Mysql database in a remote machine with a RTT= 0.187 ms, the total call throughput decreases by almost 60% in comparison when using the database in the same machine where OpenSER is installed.
Why is this happening? How can improve this behavior? I would like to have the database in a separate machine.
Thanks,
JB74
_________________________________________________________________
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Hi,
I'm trying to fetch a document from the openxcap server with a normal
browser.
My openxcap configuration uses the database backend and I created a db
table equal to the xcap table used in OpenSER.
For testing I inserted some data but I'm not sure what the fields
"source" and "doc-uri" should contain. Because whenever I try to fetch a
document
with my webbrowser I receive a 404 Not Found. OpenXCAP doesn't show any
logs...
Thanks a lot for your help!
Best regards,
Reinhold
Hi
I am trying to configure openser for uppser registration, so far I am relaying the registration after modifying contact in the registration of openser IP address but I want to save the contact only if the registration is sucessfull from upper registration server. If I use "save_noreply" its save the location no matter if the UA is registered with upper registration server
any idea
thanks in advance
---------------------------------
Never miss a thing. Make Yahoo your homepage.
Any one interested in helping with OpenSER 1.3?
Compiled openser 1.3 from svn, installed new test db from 1.3 svn db scripts
(tables should have right structure).
when a phone tries to register it hits this script section.
if (!www_authorize("", "subscriber")) {
xlog("L_INFO", "user didn't send credentials: f-uri <$fu>
r-uri <$ru>\n");
www_challenge("", "0");
exit;
};
xlog("L_INFO", "user registered: f-uri <$fu>\n");
save("location");
exit;
nothing is actually saving in location table.
I sometimes get extra messages in the log file:
core:db_print_set: Error in snprintf
usrloc:db_update_ucontact: updating database failed
usrloc:wb_timer: updating contact in db failed
I am able to use openserctl, from the openser server with all the same db
setting as the openser.cfg file has, to make changes so It "can't" be
permissions.
It is able to verify with the www_authorize that we should save it.
just not able to do the actual save.
I'm using mysql 5.x (which is also serving up the production openser
1.2-svnserver with a diff db) and openser -V renders:
version: openser 1.3.0-pre1-notls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 3015 2007-11-05 13:08:50Z henningw $
main.c compiled on 14:57:42 Nov 5 2007 with gcc 4.1.1
Thanks
Dave
Hi,
I seem to remember reading about this problem before, but couldn't find
any pointers by searching...
On the following setup with Openser 1.1:
UA -> Openser/Mediaproxy -> PSTN-Gateway
most calls pass just fine and cancel fine, but some don't stop properly.
If the UA cancels the call the cancel goes:
gateway ip= 1.2.3.204
proxy ip= 1.2.3.201
ua ip = 81.1.2.223
U 81.1.2.223:5071 -> 1.2.3.201:5060
CANCEL sip:0475711111@sipc.example.com SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.11:5071;branch=z9hG4bK-2f6ae216.
From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0.
To: <sip:0475711111@sipc.example.com>.
U 1.2.3.201:5060 -> 127.0.0.1:5060
CANCEL sip:0475711111@sipc.example.com SIP/2.0.
Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>.
Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0.
Via: SIP/2.0/UDP
192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216.
From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0.
To: <sip:0475711111@sipc.example.com>.
I 127.0.0.1 -> 1.2.3.201 3:3
....E..o..@.@...Y............[h.CANCEL sip:0475711111@sipc.example.com
SIP/2.0.
Record-Route: <sip:1.2.3.201;lr=on;ftag=a27b4fe18ca60f4ao0>.
Via: SIP/2.0/UDP 1.2.3.201;branch=z9hG4bK7f9f.b68d8185.0.
Via: SIP/2.0/UDP
192.168.1.11:5071;received=81.1.2.223;branch=z9hG4bK-2f6ae216.
From: 002 <sip:101211111@sipc.example.com>;tag=a27b4fe18ca60f4ao0.
To: <sip:0475711111@sipc.example.com>.
And then it keeps looping between 127.0.0.1 and 1.2.3.201.
For some reason openser sends the CANCEL to the loopback interface instead
of on to the gateway.
Is this a recognizable problem or should I post a full trace and config
file?
Thanks for looking,
Richard.
Hi everyone,
I'm new to WeSIP, so I've got stuck pretty fast, as expected :)
I got WeSIP running on OpenSER 1.2.0. I'm trying to run a HelloWorld sip
servlet to test WeSIP, which responses a 200 OK and a MESSAGE request
saying "Hello, World", to any incoming request to a
sip:greetings@amd-openims.test URI. I mounted my sar file, and installed
it in wesip webapps with the web manager application, and got WeSIP
properly connected to OpenSER' seas. Openser is configured to route any
MESSAGE request to WeSIP AS. But when I try to send a message to
sip:greetings@amd-openims.test from a client logged as
alice(a)amd-openims.test, WeSIP console displays this:
********************************************************************
SeasMessageEvent [SeasRouter] - New message received, from
138.4.7.163:6060 to 138.4.7.163:5080 over UDP
SeasTransactionEvent [SeasRouter] - hash_index = 54974,label=620758046
flags=0
ExpressMessageChannel [ExpressMChannel[0]] - peerAddress =
138.4.7.163/6060
ExpressMessageChannel [ExpressMChannel[0]] - About to process MESSAGE
sip:greetings@amd-openims.test SIP/2.0
SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing
processor
SipProcessor [ExpressMChannel[0]] - An incoming message is being
assigned
SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<<
MESSAGE sip:greetings@amd-openims.test SIP/2.0
Route:
<sip:138.4.7.163:5080;lr=on>,<sip:iscmark@scscf.amd-openims.test:6060;lr=on;s=1;h=0;d=0>
Via: SIP/2.0/UDP
138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP
138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP
138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926
Max-Forwards: 14
To: "Servicio HelloWorld" <sip:greetings@amd-openims.test>
From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08
Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI.
CSeq: 2 MESSAGE
Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER
Content-Type: application/unknown?
User-Agent: X-Lite release 1011s stamp 41150
P-Charging-Vector:
icid-value="P-CSCFabcd4739818c00000007";icid-generated-at=138.4.7.163;orig-ioi="amd-openims.test"
Content-Length: 262
SeasHashTable [SipProcessor[4]] - Inserting hashIdx:54974
label:620758046 into hashTable
SipProcessor [SipProcessor[4]] - Transaction [z9hG4bKeb6d.ff7360e1.0]
is new Transaction?? --> true
EngineSipMapper [SipProcessor[4]] - Request from 138.4.7.163
mapped to: ConvergedEngine[WeSIP_engine].StandardHttpHost[localhost]
ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost
ConvergedHostValve [SipProcessor[4]] - SipSession not found for
requestMESSAGE, trying match against Rules
ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched
this Request.
SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP
138.4.7.163:6060;branch=z9hG4bKeb6d.ff7360e1.0,SIP/2.0/UDP
138.4.7.163:4060;branch=z9hG4bKeb6d.000ef08.0,SIP/2.0/UDP
138.4.7.149:63926;branch=z9hG4bK-d87543-b370c6252d6ad226-1--d87543-;rport=63926
Max-Forwards: 14
To: "Servicio HelloWorld" <sip:greetings@amd-openims.test>
From: "alice" <sip:alice@amd-openims.test>;tag=737cbe08
Call-ID: Y2FiYmY0MmM2NjFhYzU5NWUzNTE2MmM5MWY2ZjdhMGI.
CSeq: 2 MESSAGE
Content-Length: 0
LingerList [SipProcessor[4]] - scheduled transaction
z9hG4bKeb6d.ff7360e1.0 for deletion
ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false
SipConnector [SipProcessor[4]] - recycle: Recycling processor
SipProcessor[4]
ActionRouter$ActionSender [ActionSender] - Writing 509 bytes into
Action Socket
SeasHashTable [ExpressMChannel[0]] - Removing hashIdx:54974
label:620758046 from hashTable
SeasHashTable [ExpressMChannel[0]] - Transaction with
id:z9hG4bKeb6d.ff7360e1.0, hash_idx:54974, label:620758046 removed from
the Seas Hash Table
ExpressMessageChannel [ExpressMChannel[0]] - Done processing MESSAGE
sip:greetings@amd-openims.test SIP/2.0
********************************************************************
And, as I feared, got a 481 response in my client.
I don't know what I'm doing wrong. I wonder it could be a misconfiguration
in the WeSIP server.xml, or maybe something wrong in my servlet.
I attach you in a zip my HelloWorld Servlet, openser.cfg file, and wesip
server.xml file.
Thanks in advance,
Cheers,
--
Alvaro
I have Openser 1.1.1 installed and working fine for long time.
But now when I restart the service openser with no changes made in
configuration this doesn't start. I have to restart many times to get the
service up again.
The error that appear in logs is the following:
Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR: init_io_wait:
could not alloc epoll array
Nov 15 11:32:06 prxc /usr/local/sbin/openser[22658]: ERROR:
tcp_receive_loop: exiting...
Do you know why this is happening?
I appreciate your help
Thanks in advanced
Ariadne
Hi,
Is it possible to change q parameter from Client side?
I'm using xlite and Sjphone, but I don't be able to change this value so,
Proxy, by default, set as 1 every q-value.
any clue?
Regards,
daniel
--
Daniel Grotti
________________________
e-mail : d.grotti(a)gmail.com
Hi, I will connect my OpenSer with a Nortel (SIP to PSTN gateway) for
incoming/outgoings call to/from PSTN.
I have a doubt: if an OpenSer user (200(a)mydomain.org) receives an incoming
call from gateway, could he REFER the call to 201(a)mydomain.org?
This is: do ITSP gateways allow REFER to SIP users subcribers to my OpenSer?
I understand they could not accept REFER's to PSTN number because accounting
issues (gateway UAC receving a REFER and calling to PSTN?).
Thanks for any explanation.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es