Hi. I am using JAIN-SIP to create a SIP message client.
When trying to create Message, Address and Header factories using
SipFactory, my program throws a PeerUnavailableException. This also occurs
when I try to create a SIP stack- I get the following exception:
javax.sip.PeerUnavailableException: The Peer SIP Stack:
gov.nist.javax.sip.SipStackImpl could not be instantiated. Ensure the Path
Name has been set.
I have set the path name to gov.nist. Does anyone know why I am getting
this exception even though I have set the SIP stack path? I am using Java
1.5. Maybe because the JAR file has been compiled with Java 1.4.2 the
byte code is incompatible? Anyway I don't know. If anybody has any ideas
about this please let me know.
Thanks,
Graham Forsyth
Graham Forsyth
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Hello List,
I am trying to run ser with rtpproxy.
I am getting this:
Feb 24 11:55:22 perch /usr/sbin/ser[31012]: ERROR: send_rtpp_command:
can't connect to RTP proxy
I have only
loadmodule "/usr/lib/ser/modules/nathelper.so"
in my ser.cfg, no other nathelper function.
I did serch in list and found 'it was answered so many times'
but again bad luck for me.
Could anybody post me the answer. I will really appreciate.
Thank,
Mohammad
Hi
Anyone using t_on_reply actually debugged the sonfig to see if it
actually goes there, I am debugging the config, and although I seem to
be going to t_on_reply(1) it doesnt seem to get there, cause the xlog
statement in that route doesnt display, I know this is a bit of a open
question...:-)
iqbal
Hi
I was just debugging my ser.cfg, and wanted to know what happens on a
break command. I presume that the break is there to stop you falling
through, but where do you jump to
EG
other code
route {
blah blah blah
route [2];
}
route[2] {
if (a=b) {do something}
else {
break;}
}
route [3] { }
In route two what happens a) when a=b and b) when the break is set.
in a) does it jump back to the start of ser.cfg and process again, or
jump back to the line after what called it, and on the "break" again
does it jump back to the calling command
Iqbal
Dear ALL:
I use Paul's ser.cfg for 0.9.0 post last year and wanna implement a
call busy forward.
I think Paul is busy on testing 'Best Practice', so he doesn't answer
too much mail.
I try to make a call from UA1 to UA2, and UA2 is busy.
So the call will forward to another Trunk 3.
Ex: UA1 ==> UA2(busy) ==(failure_route[1])==> Trunk 3
PS: UA1 and UA2 behind different NATs.
But when I watch the log, I find that miss the following log
(diff with call from UA1 to UA2 and directly forward to Trunk 3
Ex: UA1 ==> UA2(callfwd) ==> Trunk 3
When I make a callfwd call, the call will run route[3] but not failure_route[1].
UA1==>UA2(callfwd)==>route[2]==>Trunk 3
route[2] ====> SIP to SIP call
route[3] ====> SIP to PSTN call
failure_route[1] ====> NoAnswer or Busy Forward logic
And I make a fwdbusy call, the call will run route[2] then go to
failure_route[1], and return to route[3].
UA1==>UA2(fwdbusy)==>failure_route[1]==>route[3] =XXX=> Trunk 3 ????
Why does the method be failed? Do I must "end_media_session()" before
start a busy call?
How can I modify it?
My snippet ser.cfg :
--------------------------------------------------------------------------------------------------------------
route[3] {
log(1, "SER: International Call Off-Net section route(3)\n");
# All Domestic Calls Go To CISCO
if (method=="INVITE") {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
break;
} else if (!check_from()) {
log(1, "Spoofed SIP call attempt");
sl_send_reply("403", "Use From=ID");
break;
} else if (!(is_from_local() || is_uri_host_local())) {
sl_send_reply("403", "Please register to use our service");
break;
};
# enable caller id blocking for PSTN calls
if (isflagset(25)) {
append_rpid_hf();
};
};
# SIP->PSTN calls get 45 seconds to timeout
log(1, "SER: Connecting to PSTN.....\n");
avp_write("i:45", "inv_timeout");
rewritehost("xxx.xxx.190.243"); #### A Trunking Gateway
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
break;
};
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (isflagset(31)) { # is voice mail?
t_on_failure("1");
};
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
failure_route[1] {
log(1, "SER: Failure Route section failure_route(1)\n");
# if caller hung up then don't sent to voicemail
if (t_check_status("487")) {
break;
};
if (isflagset(26) && t_check_status("486")) {
# forward busy is flag 26
if (avp_pushto("$ruri", "s:fwdbusy")) {
log(1, "SER: fork to fwdbusy\n");
avp_delete("s:fwdbusy");
append_branch();
resetflag(26);
# test for domestic PSTN gateway
if (uri=~"^sip:0[0-9]{9}@") {
# if (avp_check("$fwd_busy_type", "eq/dom/i")) {
# test for domestic PSTN gateway
log(1, "SER: Busy Failure and Jump to
route(3)\n");
route(3);
} else if (uri=~"^sip:002[1-9][0-9]*@") {
# } else if (avp_check("$fwd_busy_type", "eq/int/i")) {
# test for international PSTN gateway
log(1, "SER: Busy Failure and Jump to
route(6)\n");
route(6);
} else {
# default to sip call
log(1, "SER: Busy Failure and Jump to
route(2)\n");
route(2);
};
break;
};
};
# here we can have either voicemail __OR__ forward no answer
if (isflagset(27) && t_check_status("408")) {
# forward no answer is flag 27
if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
avp_pushto("$ruri", "s:fwdnoanswer");
log(1, "SER: fork to fwdnoanswer\n");
avp_delete("s:fwdnoanswer");
append_branch();
resetflag(27);
if (uri=~"^sip:0[0-9]{9}@") {
# if (avp_check("$fwd_no_answer_type", "eq/dom/i")) {
# test for domestic PSTN gateway
log(1, "SER: No Answer Failure and Jump
to route(3)\n");
route(3);
} else if (uri=~"^sip:002[1-9][0-9]*@") {
# } else if (avp_check("$fwd_no_answer_type",
"eq/int/i")) {
# test for international PSTN gateway
log(1, "SER: No Answer Failure and Jump
to route(6)\n");
route(6);
} else {
# default to sip call
log(1, "SER: No Answer Failure and Jump
to route(2)\n");
route(2);
};
break;
};
} else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) {
avp_delete("$voicemail");
log(1, "SER: No Answer Failure and Jump to route(4)\n");
route(4);
break;
};
}
In SysLog:
------------------------------------------------------------------------------------------------------
proxydispatcher[13265]: command request
06DC9503-1938-4D9D-8628-E648F112BEC5(a)192.168.11.7
192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local
xxx.xxx.190.243 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag:
Feb 24 00:59:21 ser proxydispatcher[13265]: domain ser.xxx.net.tw
doesn't define any mediaproxy.
Feb 24 00:59:21 ser proxydispatcher[13265]: will use default
mediaproxy for this call.
Feb 24 00:59:21 ser mediaproxy[1087]: command request
06DC9503-1938-4D9D-8628-E648F112BEC5(a)192.168.11.7
192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local
xxx.xxx.190.243 remote X-PRO=20build=201082
info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag:,dispatcher
Feb 24 00:59:21 ser mediaproxy[1087]: command execution time: 0.35 ms
Feb 24 00:59:21 ser proxydispatcher[13265]: forwarding to mediaproxy
on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35030'
Feb 24 00:59:21 ser proxydispatcher[13265]: command execution time: 311.69 ms
Hello Iqbal,
Wednesday, February 23, 2005, 11:33:15 AM, you wrote:
I> Hi Mike
I> For those of us who are using asterisk, is it a replacement for the b2b
I> part of asterisk, or a add-on to asterisk to make the b2b part easier. I
I> am using asterisk for voicemail, however I need to now start to look at
I> the billing, I was just going to pas the stream via asterisk then hit
I> pstn and use the rate-engine and cdr modules for asterisk, how/where
I> does this fit in
I> tks
I> Iqbal
It's add-on that will allow to make b2bua from asterisk box, all
asterisk features are unchanged and you can use asterisk same way as
you do. All accounting and authorization is performed using radius.
P.S. Please use b2bua maillist (https://lists.berlios.de/mailman/listinfo/b2bua-users)
for questions. I think serusers list is bad place for it.
--
Best regards,
~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
Mike Tkachuk, ph:380-3433-47067
YES ISP, fx:380-3433-47067
Valova 17, mike|a|yes.net.ua
Kolomyia, www.yes.net.ua
Ukraine 78200 FWD: 66518
24.02.2005
ICQ# 57698805
MSN: mike_tkachuk|a|hotmail.com
~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
I have some issues with DNS mapping too. I want to have a sip user with
sip:user1@testdomain.com. In my location table, I however stored
sip:user1@machine1.testdomain.com as machine1.testdomain.com maps to the
actual IP address (I have such an entry in /etc/hosts). Hence although a DNS
SRV lookup is performed, it tries to contact
_sip._udp.machine1.testdomain.com, which does not exist (I only have SRV
entry for _sip._udp.testdomain.com). I think I'm missing something, can
somebody tell me where?
Any help will be greatly appreciated.
cheers
Chiang
-----Original Message-----
From: Zeus Ng [mailto:zeus.ng@isquare.com.au]
Sent: 24 February 2005 09:04
To: 'Gandhi, Chetan'
Cc: serusers(a)lists.iptel.org
Subject: RE: [Serusers] DNS SRV lookup issue
Are you absolutely sure the contact address is stored as user(a)testdomain.com
and not user(a)some.ip.address in the location table? I'm sure t_relay() will
do SRV lookup, with one condition. The URI should be in the form
sip:user@testdomain.com, with no port specified. If the URI does have the
port included like sip:user@testdomain.com:5060, then SER will not do SRV
lookup.
Most UAs I saw use IP address in contact header and that means no SRV
lookup. Better check yours with serctl ul show.
You can verify the SRV lookup with the following script.
loadmodule "/usr/local/lib/ser/modules/tm.so"
route{
rewritehost("stupid.domain");
t_relay();
}
The ethereal output will show SRV lookup.
Zeus
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Gandhi, Chetan
> Sent: Wednesday, 23 February 2005 9:15 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] DNS SRV lookup issue
>
>
> Hi All,
>
>
>
> I am facing problem to get SER to do a SRV lookup. I have the
> following scenario.
>
>
>
> Step 1:
>
> SIP UA1 registers with SER (testdomain.com) with contact
> address user(a)testdomain.com
>
> SIP UA2 registers with SER (testdomain.com) with contact
> address user(a)testdomain.com
>
>
>
> Step 2:
>
> SIP UA3 calls user(a)testdomain.com
>
>
>
> Problem
>
> I would expect SER to do a DNS SRV lookup on testdomain.com
> and get sip1.testdomain.com and sip2.testdomain.com.
> Unfortunately, SER only does A and AAAA query. (verified via
> ethereal) and does not do SRV lookup. How can I make it do
> SRV lookup ? I am using the default ser.cfg file.
>
>
>
>
>
> DNS entries
>
> _sip._udp.testdomain.com. 3600 IN
> SRV 0 1 5060 sip1.testdomain.com.
>
> _sip._udp.testdomain.com. 3600 IN
> SRV 0 1 5060 sip2.testdomain.com.
>
>
>
> I am trying to do load balancing but the SRV lookup failure
> has got me stuck. Any pointers would be really appreciated.
>
>
>
> Thanks,
>
> Chetan
>
>
_______________________________________________
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serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hello list:
I was away for little while from playing with ser.
I just switched from 0.8.14 to 0.9.0
I am getting these, when I tried to start ser.
My main concern is "submit_query(): Unknown column 'table_version' in
'field list'"
Anybody out there can help, please.
Feb 22 18:29:46 perch /usr/sbin/ser[21332]: Maxfwd module- initializing
Feb 22 18:29:47 perch /usr/sbin/ser[21332]: submit_query(): Unknown
column 'table_version' in 'field list'
Feb 22 18:29:47 perch /usr/sbin/ser[21332]: submit_query(): Error while
submitting query
Feb 22 18:29:47 perch /usr/sbin/ser[21332]: table_version(): Error in
db_query
Feb 22 18:29:47 perch /usr/sbin/ser[21332]: register_udomain(): Error
while querying table version
Feb 22 18:29:47 perch /usr/sbin/ser[21332]: domain_fixup(): Error while
registering domain
Thanks
MOhammad
Hi
I suposse you ind the solution, but if not the problem is in /etc/init.d/ser
You need to change this line
ser=/usr/sbin/ser
for this one
ser=/usr/local/sbin/ser
Cya
Mail at Cannes wrote:
> After mysql setup looks ok
>
> After ser.cfg setup believed ok
>
>
>
> When running "/etc/rc.d/init.d/ser restart"
>
> I get error message:-
>
>
>
> Stopping ser:
>
> Starting ser: execvp: No such file or directory
>
>
> [FAILED]
>
>
>
> Please can anyone give advice to resolve
>
> Thanks
>
> Chris
>
>
> ---
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> Checked by AVG anti-virus system (http://www.grisoft.com
<http://www.grisoft.com/> ).
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>
>
Hallo....
I have made a SER(SIP express router) server. i have try it in local network and it work. the client can call each other. when i try the server use IP public. a have a problem. i try using 2 client.one client uses static ip and other client uses ip public. local client can call and comunicate with the public client. but when public client call local client, it doesn't work. there some error message "408 : request time out". for information local client was under a firewall. Can you tell me the couse of error? is this because of the firewall? or there some script or other setting that must be added to the server.?
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