Hi,
This is Rakib. I install SER on my Fedora core 2 server. Now i want to know
how can i configure to dial on mobile (for Bangladesh e.g 88-0-152377618)
and pstn.
Greetings
Rakib
I was running an older version of SER and decided to upgrade my
installation. I downloaded a solaris 10 installation and did the pkgadd
-d to install it. Everything seems to have worked, but when I try and
start ser using serctl start I get the following response:
Starting SER : PID file /var/run/ser.pid does not exist -- SER start
failed
I can't seem to get past this point. Why does it care if the ser.pid
file exists since it is supposed to be starting the application? I
would appreciate any suggestions?
Hi, I have this error,
undefined symbol: db_free_query
I change the version of ser, 0.8.14 to ser-0.10.99-dev30-tm-timers-pa-3
what is the funcion to replace db_free_query ?
best regards
Hi Don,
Thank you for your reply. How can I check using commands that the thread is still in session? What are
the commands in the scripts that close the rtp session? Please advice. Thanks
Regards,
Nicky
----- Original Message -----
From: don.fletcher(a)att.net
To: Nicky
Sent: Tuesday, February 21, 2006 2:47 AM
Subject: Re: [Sems] Sems Thread Memory
Nicky,
Probably the RTP streams are not being closed down properly at the end of a call so that the threads remain in existence. Look at the media streams with ethereal. Look closely at call disconnect. Be sure all threads are closed.
don
-------------- Original message from "Nicky" <nicky(a)caliber.com.sg>: --------------
Dear All,
I am having a serious problem, my thread memory size is growing to up to 2GB which applies to SEMS, SER and Mysqld thread.
Please help. I am testing with about 10 people calling in at the same time. What shall I do? :(
Nicky
----- Original Message -----
From: Nicky
To: serusers(a)lists.iptel.org ; sems(a)lists.iptel.org
Sent: Saturday, October 29, 2005 11:44 AM
Subject: pthread create failed with code 11
Hi Support,
I have experienced this error, please advice what can be the caused. Thanks.
ERROR: start (AmThread.cpp:85): pthread create failed with code 11
Regards,
Nicky
What about loose_route ? ReINVITE must be catched by sloose_route and...
for example in my config almost in the beginning, before all kind of
auth checks:
if (loose_route()) {
if(method == "INVITE") {
xlog("L_DBG", "%ci: route, loose_route said yes
for INVITE msg\n");
route(1);
break;
} else {
xlog("L_DBG", "%ci: route, loose_route said yes
for non INVITE msg\n");
t_relay();
break;
}
};
Then on route(1) there is auth.. (it was before in main route) I do
usual nat stuff and forward the call to whenever Route: field pinted:
t_on_reply("1");
if (! loose_route()) {
if ((!lookup("location") && method == "INVITE" && uri ==
myself) || uri == myself) {
sl_send_reply("404", "Not Found");
break;
};
}
if(method == "INVITE") {
/* Handle re-INVITEs */
if (force_rtp_proxy("l")) {
t_on_reply("2");
} else {
if(force_rtp_proxy(""))
t_on_reply("2");
};
};
if (!t_relay()) {
sl_reply_error();
};
So, ReINVITE will be catched in forst loose_route if and bypass auth
section will be sent to route(1) where it will not do lookup(location)
but send wherever route field pointed.
I am wrong ? :)
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Pat wang
Sent: Tuesday, February 21, 2006 10:36 AM
To: klaus.mailinglists(a)pernau.at
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Authentication of ReInvite
Hi Klaus,
The /var/log/messages does not show any thing but here is the ngrep of
the
Re-Invite. I couldn't find anything wrong in the ACK for 407. Do you
have
any other thoughts?
Thanks,
Pat
U 192.10.1.13:5060 -> 192.10.1.2:5060
INVITE sip:2101@192.10.1.11:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805.
From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695.
To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1.
Route:
<sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr=
on>.
CSeq: 2 INVITE.
Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13.
Contact: <sip:2103@192.10.1.13:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0.
Supported: 100rel,timer,replace.
Allow-Events: talk,hold,conference.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 197.
.
v=0.
o=- 1137681804 1137681804 IN IP4 192.10.1.13.
s=Polycom IP Phone.
c=IN IP4 192.10.1.13.
t=0 0.
m=audio 2252 RTP/AVP 0 101.
a=sendonly.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
U 192.10.1.2:5060 -> 192.10.1.13:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805.
From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695.
To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1.
CSeq: 2 INVITE.
Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13.
Proxy-Authenticate: Digest realm="test.com",
nonce="43fb2e105733df37da780239bc94922da01a4177".
Server: Sip EXpress router (0.9.6 (i386/linux)).
Content-Length: 0.
Warning: 392 192.10.1.2:5060 "Noisy feedback tells: pid=26457
req_src_ip=192.10.1.13 req_src_port=5060
in_uri=sip:2101@192.10.1.11:5060
out_uri=sip:2101@192.10.1.11:5060 via_cnt==1".
.
U 192.10.1.13:5060 -> 192.10.1.2:5060
ACK sip:2101@192.10.1.11:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805.
From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695.
To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1.
Route:
<sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr=
on>.
CSeq: 2 ACK.
Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13.
Contact: <sip:2103@192.10.1.13:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0.
Max-Forwards: 70.
Content-Length: 0.
.
U 192.10.1.2:5060 -> 192.10.1.11:5060
ACK sip:2101@192.10.1.11:5060 SIP/2.0.
Record-Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>.
Via: SIP/2.0/UDP 192.10.1.2;branch=0.
Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805.
From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695.
To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1.
CSeq: 2 ACK.
Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13.
Contact: <sip:2103@192.10.1.13:5060>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY,
PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0.
Max-Forwards: 16.
Content-Length: 0.
P-hint: rr-enforced.
.
>From: Klaus Darilion <klaus.mailinglists(a)pernau.at>
>To: Pat wang <wangyu39(a)hotmail.com>
>CC: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Authentication of ReInvite
>Date: Tue, 21 Feb 2006 13:04:59 +0100
>
>Pat wang wrote:
>
>>Morning everyone,
>>
>>Has anyone tried authentication of Re-INVITE on SER? When I add the
>>proxy_challenge to the ReInvite in the SER configure file, the ACK for
407
>>from the client is porxied by SER to the other side. The signalling
looks
>>like:
>>
>>caller----------------------SER---------------------Callee
>>------------------ normal call setup--------------------
>>--------------------------blha blha-----------------------
>>-------------------------------------------------------------
>>---call hold Re-Invite--->
>><--- 407 challenge------
>>--------ACK----------------->
>> -------------ACK---------> *** This
is
>>not to be proxied!
>>----ReInvite--------------->
>>----------------------etc etc............
>>
>>Anyone seen this while doing similar thing on SER? How can I stop SER
>>proxing this ACK just like the way SER treat the original Invite with
>>authentiacion?
>>
>>Here is the authentication part in the loose route block of my SER
config
>>file:
>>
>> if (method=="INVITE") {
>> if (!proxy_authorize("test.com", "subscriber")) {
>> proxy_challenge( "test.com", "0");
>> break;
>> };
>> };
>
>AFAIK, proxy challenge sends a stateless reply. ACK to stateless
replies
>should be absorbed by ser immediately without entering the ser.cfg
routing
>script. Thus, probably ser can't detect this ACK as stateless.
>
>Please watch syslog (debug=4 and "tail -f /var/log/syslog|grep -v qm_")
>and verify why ser does nto detect a "stateless" ACK.
>
>Also verifiy the ACK if all the tags are correctly copied from 40x to
the
>ACK. Maybe this is cause by the branch=0 bug?
>
>regards
>klaus
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hello,
the development version of textops module has a new set of functions
that work on the body of the SIP messages.
- search_body() - search a regular expression within the message's body
- search_append_body() - search a regular expression within the message's body and append a text after it
- replace_body() - replace a regular expression match with a text
- replace_body_all() - replace all regular expression matches with a text
- subst_body() - sed/perl like subst operation over the message's body
More details about these functions at:
http://openser.org/docs/modules/1.1.x/textops.html
Cheers,
Daniel
Morning everyone,
Has anyone tried authentication of Re-INVITE on SER? When I add the
proxy_challenge to the ReInvite in the SER configure file, the ACK for 407
from the client is porxied by SER to the other side. The signalling looks
like:
caller----------------------SER---------------------Callee
------------------ normal call setup--------------------
--------------------------blha blha-----------------------
-------------------------------------------------------------
---call hold Re-Invite--->
<--- 407 challenge------
--------ACK----------------->
-------------ACK---------> *** This is not
to be proxied!
----ReInvite--------------->
----------------------etc etc............
Anyone seen this while doing similar thing on SER? How can I stop SER
proxing this ACK just like the way SER treat the original Invite with
authentiacion?
Here is the authentication part in the loose route block of my SER config
file:
if (method=="INVITE") {
if (!proxy_authorize("test.com", "subscriber")) {
proxy_challenge( "test.com", "0");
break;
};
};
Thanks,
Pat
If I have a prefix 1100 and I send call to 11000202, how can I use
ENUM this?
If I have the line below, I would need to put username but I don´t want
this:
$ORIGIN 0.2.0.0.0.1.1.e164.arpa.
2 IN NAPTR 1 10 "u" "E2u+sip" "!^.*$!sip:10.10.10.1<r(a)10.10.10.1>;
sip:john@10.10.10.2!" .
Suppose, I send to 11000202 or 11000203 I want send call to 10.10.10.2 but
I want to use the e164 original. If have a lot of users I would need to put
all the users in ENUM, but I don´t want this.
How can I do this?
