date:20101013

[OpenSIPS-Users] How opensis can manage different ports.

2010-10-13 Thread Marcella




KINGSSQUEEN





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] [OpenSIPS LiveDVD]Register OK but call failed:prompt "420 invalid destination"

2010-10-13 Thread Marcella




KINGSSQUEEN





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] (no subject)

2010-10-13 Thread Marcella




KINGSSQUEEN





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Static Routes in OpenSIPS proxy

2010-10-13 Thread andrea


Hi,

i'm a beginner of OpenSIPS, i've installed 1.6 version using standard
configuration.
I would like to know i how add static routes (or SIP trunk) in a proxy
server based OpenSIPS.
In particular,  Can you suggest me a modification in the configuration file
so that this proxy can route the calls incoming from an IP-PABX sip to
phones registerd to others IP-PABX via SIP trunks?

Thanks in advance.
Andrea 
-- 
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Static-Routes-in-OpenSIPS-proxy-tp5623260p5623260.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Static Routes in OpenSIPS proxy

2010-10-13 Thread andrea


Hi,

i'm a beginner of OpenSIPS, i've installed 1.6 version using standard
configuration.
I would like to know i how add static routes (or SIP trunk) in a proxy
server based OpenSIPS.
In particular,  Can you suggest me a modification in the configuration file
so that this proxy can route the calls incoming from an IP-PABX sip to
phones registerd to others IP-PABX via SIP trunks?

Thanks in advance.
Andrea 
-- 
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Static-Routes-in-OpenSIPS-proxy-tp5623248p5623248.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Core dump in svn head

2010-10-13 Thread Dave May

Yes, I do set a branch_route to strip RPID header and strip/set the
P-Asserted-Identity header.

I'm using drouting, and that portion of the failure_route is based on
the code from Flavio's book (with the addition of e164 mangling,
t_on_branch to handle the header stuff mentioned above, and t_on_failure
to handle multiple failures):



if(isflagset(10)){
if (!t_check_status("408|[56][0-9][0-9]")) {
# this is not a GW failure
exit;
}

if (use_next_gw()) {
xlog("$ci\t$C(rc)$si $rm$C(xx)
$C(xr)failure_route[1]>$C(xx)
$C(xr)use_next_gw()=$C(rg)$ru$C(xx)\n");

###  Need to duplicate e164 check from original
do_routing() call
###  Want to do this in branch_route, but not
allowed to use is_uri_user_e164() there
###
#
# Does the gateway entry request e164 format?
if ($avp(s:dr_gw_attrs) != "") &&
($(avp(s:dr_gw_attrs){param.value,format})=="e164") {

xlog("$ci\t$C(rc)$si $rm$C(xx)
failure_route[1]> Gateway requires e164 format\n");

# Check and fix RURI
if (is_uri_user_e164("$ru")) {
xlog("$ci\t$C(rc)$si $rm$C(xx)
failure_route[1]> $ru is in e164 format\n");
} else {
xlog("$ci\t$C(rc)$si $rm$C(xx)
failure_route[1]> $ru is not in e164 format\n");
dp_translate("3");
}
}

# Need to ensure that failures on the failure
are also handled
t_on_branch("1");
t_on_failure("1");

xlog("$ci\t$C(rc)$si $rm$C(xx)
failure_route[1]> t_relay() next  rU = $rU   fU=$fU
tU=$tU   du=$du\n");
setdebug(4);
t_relay();
setdebug();
xlog("$ci\t$C(rc)$si $rm$C(xx)
failure_route[1]> t_relay() complete\n");
exit;
} else {
xlog("$ci\t$C(rc)$si $rm$C(xx)
$C(xr)failure_route[1]->$C(xx) isflagset(10) $C(sr)--> use_next_gw()
FAILED!!!$C(xx)\n");

t_reply("503", "Service not available, no more
gateways");
exit;
}

Dave.

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Wednesday, October 13, 2010 4:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Core dump in svn head

Hi Dave,

In failure route, how do you add the new destination/branch ?

Also, do you have a branch route set ?

Regards,
Bogdan

Dave May wrote:
> Over the past couple weeks I have been getting occasional segfaults
just
> prior to (or perhaps in the process of) a t_relay() in my
failure_route.
> Still haven't gotten to the bottom of the root cause in my config, but
I
> was able to find and fix the symptomatic code in the
> pre_print_uac_request() function in t_fwd.c (based on Bogdan's
previous
> fix for a similar problem).
>
> Backtrace:
> #0  0xb763ba55 in memcpy () from /lib/tls/i686/cmov/libc.so.6
> #1  0xb72ede1d in pre_print_uac_request (t=0x972ddf78, branch=1,
> request=0x972cec64) at t_fwd.c:177
> #2  0xb72eeac0 in add_uac (t=0x972ddf78, request=0x7fff,
> uri=0xbf8a250c, next_hop=0xbf8a2514, path=0xb7315100, proxy=0x0) at
> t_fwd.c:401
> #3  0xb72efbc0 in t_forward_nonack (t=0x972ddf78, p_msg=0xb7314de0,
> proxy=0x0) at t_fwd.c:660
> #4  0xb72fbdbf in w_t_relay (p_msg=0xb7314de0, proxy=0x0, flags= optimized out>) at tm.c:1116
> #5  0x08057810 in do_action (a=0x819f924, msg=0xb7314de0) at
> action.c:1155
> #6  0x080557ff in run_action_list (a=0x819e7cc, msg=0xb7314de0) at
> action.c:140
> #7  0x0805981e in do_action (a=0x819fed4, msg=0xb7314de0) at
> action.c:821
> #8  0x080557ff in run_action_list (a=0x819e584, msg=0xb7314de0) at
> action.c:140
> #9  0x0805981e in do_action (a=0x81a0390, msg=0xb7314de0) at
> action.c:821
> #10 0x080557ff in run_action_list (a=0x819d588, msg=0xb7314de0) at
> action.c:140
> #11 0x0805aef1 in run_top_route (a=0x819d588, msg=0xb7314de0) at
> action.c:120
> #12 0xb7304c2c in t_should_relay_response (Trans=0x972ddf78,
> new_code=408, branch=0, should_store=0xbf8a3094,
> should_relay=0xbf8a3098, cancel_bitmap=0xbf8a30e8, reply=0x)
> at t_reply.c:621
> #13 0xb730683d in relay_reply (t=0x972ddf78, p_msg=0x,

Re: [OpenSIPS-Users] Changing Opensips port number

2010-10-13 Thread James Mbuthia

Hi Bogdan,

I set the port on opensips.cfg like:


port=5060

/* uncomment and configure the following line if you want opensips to
   bind on a specific interface/port/proto (default bind on all available) */
listen=udp:198.162.1.0:5060


I change the 5060 to another number then restart using opensipsctl. Is
there something else I have missed?

james



On 10/13/10, Bogdan-Andrei Iancu  wrote:
> Hi James,
>
> how do you set the port? via "port" param or via the "listen" param? are
> you sure you restart was effective?
>
> Regards,
> Bogdan
>
> James Mbuthia wrote:
>> Hi,
>>
>> Sorry for the newbie question. How do you change the port number in
>> opensips? I want to change it from 5060 to something else however when
>> I do so on opensips.cfg and restart opensips, Opensips doesn't seem to
>> listen on the port I have specified. Do I need to change the number
>> elsewhere?
>>
>> regards
>> james
>> 
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] DNS Failover test

2010-10-13 Thread James Mbuthia

But when does the proxy create a location record?

When the callee phone registers it gets an authenticate challenge and
after the challenge it gets a 200 Ok header.

When the caller phone calls it gets a 407 proxy challenge and after
verification gets a 100 response.

According to my understanding the proxy creates a location record for
the person being called, however this is  not entered into the
database so where is it? Below is a snippet of my invite and register
code in opensips.cfg


if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/
##if (!(method=="REGISTER") && is_from_local())  /*multidomain */
##
{
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
if (!db_check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}

consume_credentials();
## # caller authenticated
}


if (is_method("REGISTER"))
{
# authenticate the REGISTER requests (uncomment to enable auth)
if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
##
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}

if (!save("location"))
sl_reply_error();

exit;
}
}



The proxy does not save the location details under the location table
however sending an invite before the callee has been registered
results in a 405. So, where does the proxy save the callee location if
its not saving on the database? And how do I need to go around that?


On 10/13/10, Bogdan-Andrei Iancu  wrote:
> Right, and before that, when receiving a request, a proxy looks at the
> domain part of RURI to see if that domain is a SIP domain that can be
> locally handled or a it is a foreign SIP domain that need to forward to
> another SIP server.
>
> So, check if the domain RURI is a local domain - see the default
> opensips.cfg for an example how to do it.
>
> Regards,
> Bogdan
>
> Duane Larson wrote:
>>
>> I could be wrong but does your phone register with the proxy and then
>> the proxy create a location record for each user?  That would be how
>> opensips would know how to route it.
>>
>>> On Oct 13, 2010 1:04 PM, "James Mbuthia" >> > wrote:
>>>
>>> Hi guys,
>>>
>>> Am a SIP and Opensips newbie and am trying to create a proof of
>>> concept on a unique (I think) Opensips and SIPS implementation.
>>> Basically am developing a web-based SIP Phone based on PHP and using
>>> Opensips. Because of the nature of the application, users will have
>>> the same domain and just one proxy server which will process invites
>>> and registrations.  Based on my understanding of SIP and Opensip, I
>>> have developed a UAC that can send invites and registrations to a
>>> proxy server. My problem is coming when I have to route a call from
>>> one agent to another. According to RFC 3263 the proxy server needs to
>>> determine the sip server for the person receiving the call. Know if
>>> the proxy server of the person receiving the call and the person
>>> initiating the call is the same and both the caller and callee are on
>>> the same domain than how does Opensips route the message?
>>>
>>> I've worked on this the last few days and below is an excerpt from my
>>> log files. For some reason the proxy server is not able to route the
>>> message to user2 therefore it sends a message to user1 changing the
>>> SIP header from 100 to 407(Proxy authorization needed).
>>>
>>>
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
>>> DBG:tm:build_local: using FROM=>> >;tag=232523
>>>  >, TO=>> >
>>> ;tag=9360cfebad69401b0dbd073e6bd6f131.188c
>>>  >, CSEQ_N=
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
>>> incoming reply
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
>>> DBG:core:parse_msg: SIP Request:
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
>>> DBG:tm:t_should_relay_response: T_code=100, new_code=407
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
>>> DBG:core:parse_msg:  method:  
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
>>> DBG:tm:t_pick_branch: picked branch 0, code 407 (prio=501)
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
>>> DBG:core:parse_msg:  uri: >> >
>>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
>>> DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=407,
>>> flags=2
>>>
>>>
>>> Could anyone give me some insights o

[OpenSIPS-Users] Services management - question about proper module

2010-10-13 Thread Maciej Bylica

Hello.

I am planning to provide opensips with a kind of mechanism to manage
customer services/features like call-forward/VM/follow-me and so on.
It should work in following way: If $rU is provided in subscriber
table then user enabled service name is obtained from some db table.
On the basis of that value opensips should do the magic :)

The question is what kind of module is the best to follow. Is it
AVPops or maybe there is another way to achieve my goal.
What are pros and cons.

Thx in advance,
Maciej.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Serialforking failure, with lcr:parse_phostport: too many colons in udp:: 0

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Taisto,

Your problem is not timer related or how serial forking is done in 
opensips (I will comment on these in a later reply).

Right now, the quick answer to fix your problem: failure route must be 
re-armed after each branch -> this is why your failure route does not 
catches the end of the second branch. Adding a t_on_failure("1") before 
t_relay() in failure route will fix your problem.

Regards,
Bogdan



Taisto Qvist wrote:
> Hi Bogdan,
>
> I've now been trying with some tests, and I cant really get it to work,
> since the transactionlayer on the server transaction returns a 408
> back to the UAC before serial forking has ended.
> This seems a little bit related to what I once commented on a long time
> ago regarding handling of timer C and the fact that the timer c seems to
> be quite "tied" to Timer B
>
> When the fr_timer pops, (causing the CANCEL to be sent so that we can move
> on to the next serial-fork-target), the tm-layer seems store this 
> timer-pop
> as a 408 response
>
> 20:41:44 osips[4686]: DBG:tm:utimer_routine: timer 
> routine:4,tl=0xb5b6770c next=(nil), timeout=64930
> 20:41:55 osips[4686]: DBG:tm:timer_routine: timer 
> routine:1,tl=0xb5b67728 next=(nil), timeout=660
> 20:41:55 osips[4686]: DBG:tm:final_response_handler: stop retr. and 
> send CANCEL (0xb5b675c0)
> 20:41:55 osips[4686]: DBG:tm:t_should_relay_response: T_code=180, 
> new_code=408
> 20:41:55 osips[4686]: DBG:tm:t_pick_branch: picked branch 0, code 408 
> (prio=800)
>
> As the capture and log I've attached indicates, I am not able to 
> perform a three
> step serial fork. I have three Uas:es registered with 1.0, 0.9, and 
> 0.8 in q-values.
>
> First timer pop causes a CANCEL, and a new INVITE towards UAS with 
> q=0.9, but when
> it pops the second time, TM still cancels the second target, but 
> instead of continuing
> with the third, it sends a 408 towards the UAC.
>
> It might be something with my script-handling in the failure_route, so 
> here it is:
>
> failure_route[1]
> {
> if ( t_was_cancelled() )
> {
> log(2, "transaction was cancelled by UAC\n");
> }
> xlog("(lab1) - In FailureRoute: branches=$(branch(uri)[*])\n");
> if ( isflagset(1) )
> {
> log(2,"(lab1) - 3++ Received, attempting serial fork!\n");
> next_branches();
> switch ( $retcode )
> {
> case 1:
>log(2,"(lab1) - More branches left, rollOver timer 
> set.");
>$avp(s:timerC) = 12;
>setflag(1);  # Do I need this? Should I use 
> branchflags instead?
> break;
> case 2:
>log(2,"(lab1) - Last branch, timerC set to 60 sec");
>$avp(s:timerC) = 60;
> break;
>
> default:
>log(2,"(lab1) - No more serial fork targets.");
>exit;
> }
> if ( !t_relay() )
> {
> log(2,"(lab1) - Error during relay for serial fork!\n");
> }
> }
> else
> {
> log(2,"(lab1) - 3++ result. Serialforking not available.\n");
> }
>
> }
>
> When I say that it seems related to another issue I commented on a 
> long time
> ago, I am referring to the general handling of Timer C, which doesn't 
> seem to
> be a separate timer, but is reusing the timerB.
>
> When the timer pops after the normal 180 seconds, the TM layer will 
> *instantly*
> generate a 408 response on the server txn, while at the same time 
> generating
> the CANCEL attempting to terminate the client txn.
> To me, this is wrong, but maybe I am suppose to handle this in the 
> failure_route?
>
> What I would expect is that the CANCEL will cause a 487 response from 
> the UAS,
> and this will be the final response sent to the UAC.
> Also by behaving this way, we may cause a protocol violation even 
> though the risk
> is small.
>
> Once timer C pops we send the CANCEL hoping that it will cause a 487. 
> BUT, it is
> quite possible that before the cancel is received by the UAS, it sends 
> a 200 to
> the INVITE! Even IF the CANCEL receives a 2xx response, we may still 
> get a 2xx
> response to the INVITE.
> But with the current behavior of opensips, this would cause opensips 
> to proxy
> TWO final responses on the server txn, once being the initial 408 sent 
> by the
> txn on timer C timeout, and then the real/actual 2xx sent by the uas.
>
> I've also seen a similar problem with 6xx responses received on a 
> branch during
> forking.
> Opensips forwards the 6xx *before* the remaining client txns has 
> completed, and
> there is no guarantee that these client txns will all terminate with 
> 487 even
> if opensips tries to CANCEL all of them asap.
> They may still return 2xx to the invite, which would cause a 
> forwarding of both
> a 6xx and a 2xx on the server txn. This scenario is even mentioned in 
> rfc3261.
>
> So all these three problems have in common that the server txn seems 
> to be
>

Re: [OpenSIPS-Users] B2BUA Transfers Intermittently Working/Not Working (Bug?)

2010-10-13 Thread Brett Woollum

Unfortunately the mailing list won't let me send replies with my logs or config 
file since they're too big. 

You can find them on this page: http://www.woollum.com/temp/opensips.txt 

Thanks! 


Brett Woollum 
br...@woollum.com 


- Original Message - 
From: "Brett Woollum"  
To: "users-request"  
Sent: Wednesday, October 13, 2010 1:50:06 PM GMT -08:00 US/Canada Pacific 
Subject: [OpenSIPS-Users] B2BUA Transfers Intermittently Working/Not Working 
(Bug?) 


Hello Everyone, 


In an attempt to figure out the best way to build my OpenSIPS config with the 
B2BUA module included, I've started over with a very simple script implementing 
nothing but the B2BUA module (and usrloc). My goal is to allow the phones to 
place calls between them and transfer the calls to other local phones (similar 
to a PBX). 

I have the B2BUA module configured to allow the call to be transferred (REFER) 
up to 4 times. This scenario script and OpenSIPS config file work well 
depending on which leg of the call is being transferred. It appears that the 
originator of the call cannot be put back on the opposite leg of the call, as 
shown in the second example above. Another way to think of the problem is that 
the remote side cannot be transferred. 

For both of the scenarios below, I have 3 phones. One is for the Sales 
department, one is for Tech Support, and the third represents the Boss. 

Here is a scenario where I can transfer the call four times successfully: 
1. Sales places a call to Tech Support. (Sales has the "local" side of the 
call, and Tech Support has the "remote" side of the call.) 
2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
3. The Boss transfers his leg of the call back to Sales. (Sales has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
4. Sales transfers his leg of the call back to the Boss. (The Boss has the 
"local" side of the call, and Tech Support has the "remote" side of the call.) 

Here is a scenario where the process is interrupted: 
1. Sales places a call to Tech Support. (Sales has the "local" side of the 
call, and Tech Support has the "remote" side of the call.) 
2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
3. Tech Support transfers his leg of the call back to Sales. (The Boss has the 
"local" side of the call, and Sales has the "remote" side of the call.) 
**THE CALL FAILS AT THIS POINT** - The Boss's phone is left on the call with 
nobody on the other end. Tech Support's phone appears to transfer successfully. 
Nothing happens with the Sales phone. 

My OpenSIPS Server is at 1.2.3.4 in this example, and the phones are on the 
10.20.1.x subnet. There is no NAT between the devices. This appears to be no 
correlation between which devices are doing the transferring (meaning the Boss 
and Tech Support could switch roles and it still fails). I am running OpenSIPS 
1.6.3 on CentOS 5.5 x64. 

I'll send my log file, config file, and scenario file in a moment (they were 
too big to fit in one email). 

Brett Woollum 
br...@woollum.com 
___ Users mailing list 
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Dialplan problem ?

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Marcio,

The answer is:

Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:build_rule: 
references:1 , max:1

Oct 13 17:05:47 perseu /sbin/opensips[13077]: DBG:dialplan:destroy_rule: 
destroying rule with priority 1


It looks like opensips rejects the rules while loading them at startup, 
so basically you end up with no rule at runtime.

what opensips version are you using ?

Regards,
Bogdan

Marcio Veloso Antunes wrote:
>
> Hi guys,
>
> Sorry to bother you, but i can't find the problem why this is not working:
>
> Actual dialplan:
>
> r...@perseu:/etc/opensips# opensipsctl dialplan show
>
> dialplan tables
>
> id | dpid | pr | match_op | match_exp | match_len | subst_exp | 
> repl_exp | attrs
>
> +--++--+---+---+-+--+---
>
> 18 | 1 | 0 | 1 | ^00[1-9][0-9]+ | 0 | ^0(0[1-9][0-9]+) | \1 |
>
> 19 | 1 | 1 | 1 | ^00[1-9][1-9][0-9]{8} | 0 | ^0(0[1-9][1-9][0-9]{8}) | 
> \1 |
>
> (2 rows)
>
> String being tested: '002185392949'
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] B2BUA Transfers Intermittently Working/Not Working (Bug?)

2010-10-13 Thread Brett Woollum

Hello Everyone, 


In an attempt to figure out the best way to build my OpenSIPS config with the 
B2BUA module included, I've started over with a very simple script implementing 
nothing but the B2BUA module (and usrloc). My goal is to allow the phones to 
place calls between them and transfer the calls to other local phones (similar 
to a PBX). 

I have the B2BUA module configured to allow the call to be transferred (REFER) 
up to 4 times. This scenario script and OpenSIPS config file work well 
depending on which leg of the call is being transferred. It appears that the 
originator of the call cannot be put back on the opposite leg of the call, as 
shown in the second example above. Another way to think of the problem is that 
the remote side cannot be transferred. 

For both of the scenarios below, I have 3 phones. One is for the Sales 
department, one is for Tech Support, and the third represents the Boss. 

Here is a scenario where I can transfer the call four times successfully: 
1. Sales places a call to Tech Support. (Sales has the "local" side of the 
call, and Tech Support has the "remote" side of the call.) 
2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
3. The Boss transfers his leg of the call back to Sales. (Sales has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
4. Sales transfers his leg of the call back to the Boss. (The Boss has the 
"local" side of the call, and Tech Support has the "remote" side of the call.) 

Here is a scenario where the process is interrupted: 
1. Sales places a call to Tech Support. (Sales has the "local" side of the 
call, and Tech Support has the "remote" side of the call.) 
2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" 
side of the call, and Tech Support has the "remote" side of the call.) 
3. Tech Support transfers his leg of the call back to Sales. (The Boss has the 
"local" side of the call, and Sales has the "remote" side of the call.) 
**THE CALL FAILS AT THIS POINT** - The Boss's phone is left on the call with 
nobody on the other end. Tech Support's phone appears to transfer successfully. 
Nothing happens with the Sales phone. 

My OpenSIPS Server is at 1.2.3.4 in this example, and the phones are on the 
10.20.1.x subnet. There is no NAT between the devices. This appears to be no 
correlation between which devices are doing the transferring (meaning the Boss 
and Tech Support could switch roles and it still fails). I am running OpenSIPS 
1.6.3 on CentOS 5.5 x64. 

I'll send my log file, config file, and scenario file in a moment (they were 
too big to fit in one email). 

Brett Woollum 
br...@woollum.com ___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Core dump in svn head

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Dave,

In failure route, how do you add the new destination/branch ?

Also, do you have a branch route set ?

Regards,
Bogdan

Dave May wrote:
> Over the past couple weeks I have been getting occasional segfaults just
> prior to (or perhaps in the process of) a t_relay() in my failure_route.
> Still haven't gotten to the bottom of the root cause in my config, but I
> was able to find and fix the symptomatic code in the
> pre_print_uac_request() function in t_fwd.c (based on Bogdan's previous
> fix for a similar problem).
>
> Backtrace:
> #0  0xb763ba55 in memcpy () from /lib/tls/i686/cmov/libc.so.6
> #1  0xb72ede1d in pre_print_uac_request (t=0x972ddf78, branch=1,
> request=0x972cec64) at t_fwd.c:177
> #2  0xb72eeac0 in add_uac (t=0x972ddf78, request=0x7fff,
> uri=0xbf8a250c, next_hop=0xbf8a2514, path=0xb7315100, proxy=0x0) at
> t_fwd.c:401
> #3  0xb72efbc0 in t_forward_nonack (t=0x972ddf78, p_msg=0xb7314de0,
> proxy=0x0) at t_fwd.c:660
> #4  0xb72fbdbf in w_t_relay (p_msg=0xb7314de0, proxy=0x0, flags= optimized out>) at tm.c:1116
> #5  0x08057810 in do_action (a=0x819f924, msg=0xb7314de0) at
> action.c:1155
> #6  0x080557ff in run_action_list (a=0x819e7cc, msg=0xb7314de0) at
> action.c:140
> #7  0x0805981e in do_action (a=0x819fed4, msg=0xb7314de0) at
> action.c:821
> #8  0x080557ff in run_action_list (a=0x819e584, msg=0xb7314de0) at
> action.c:140
> #9  0x0805981e in do_action (a=0x81a0390, msg=0xb7314de0) at
> action.c:821
> #10 0x080557ff in run_action_list (a=0x819d588, msg=0xb7314de0) at
> action.c:140
> #11 0x0805aef1 in run_top_route (a=0x819d588, msg=0xb7314de0) at
> action.c:120
> #12 0xb7304c2c in t_should_relay_response (Trans=0x972ddf78,
> new_code=408, branch=0, should_store=0xbf8a3094,
> should_relay=0xbf8a3098, cancel_bitmap=0xbf8a30e8, reply=0x)
> at t_reply.c:621
> #13 0xb730683d in relay_reply (t=0x972ddf78, p_msg=0x, branch=0,
> msg_status=408, cancel_bitmap=0xbf8a30e8) at t_reply.c:1133
> #14 0xb72f21e5 in fake_reply (t=0x972ddf78, branch=0, code=408) at
> timer.c:259
> #15 0xb72f3dd7 in timer_routine (ticks=3285, attr=0x0) at timer.c:365
> #16 0x080c12c0 in start_timer_processes () at timer.c:325
> #17 0x0806e37e in main (argc=Cannot access memory at address 0x7fff
> ) at main.c:872
>
> Here is the patch to fix:
>
> Index: modules/tm/t_fwd.c
> ===
> --- modules/tm/t_fwd.c  (revision 7259)
> +++ modules/tm/t_fwd.c  (working copy)
> @@ -166,7 +166,7 @@
> -request->REQ_METHOD);
>  
> /* copy dst_uri into branch (after branch route possible updated
> it) */
> -   if (request->dst_uri.len) {
> +   if ( request->dst_uri.s && request->dst_uri.len>0 ) {
> t->uac[branch].duri.s =
> shm_resize(t->uac[branch].duri.s,
> request->dst_uri.len);
> if (t->uac[branch].duri.s==NULL) {
>
>
> After the code change and a recompile, the following lines appear in my
> logs, whereas previously at this spot OpenSIPS crashed:
>
> Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:core:parse_uri:
> uri too short: <> (-1) 
> Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:tm:uri2proxy:
> bad_uri: (null) 
> Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]:
> ERROR:tm:t_forward_nonack: failure to add branches 
> Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:tm:w_t_relay:
> t_forward_nonack failed
>
> Dave.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] DNS Failover test

2010-10-13 Thread Bogdan-Andrei Iancu

Right, and before that, when receiving a request, a proxy looks at the 
domain part of RURI to see if that domain is a SIP domain that can be 
locally handled or a it is a foreign SIP domain that need to forward to 
another SIP server.

So, check if the domain RURI is a local domain - see the default 
opensips.cfg for an example how to do it.

Regards,
Bogdan

Duane Larson wrote:
>
> I could be wrong but does your phone register with the proxy and then 
> the proxy create a location record for each user?  That would be how 
> opensips would know how to route it.
>
>> On Oct 13, 2010 1:04 PM, "James Mbuthia" > > wrote:
>>
>> Hi guys,
>>
>> Am a SIP and Opensips newbie and am trying to create a proof of 
>> concept on a unique (I think) Opensips and SIPS implementation. 
>> Basically am developing a web-based SIP Phone based on PHP and using 
>> Opensips. Because of the nature of the application, users will have 
>> the same domain and just one proxy server which will process invites 
>> and registrations.  Based on my understanding of SIP and Opensip, I 
>> have developed a UAC that can send invites and registrations to a 
>> proxy server. My problem is coming when I have to route a call from 
>> one agent to another. According to RFC 3263 the proxy server needs to 
>> determine the sip server for the person receiving the call. Know if 
>> the proxy server of the person receiving the call and the person 
>> initiating the call is the same and both the caller and callee are on 
>> the same domain than how does Opensips route the message? 
>>
>> I've worked on this the last few days and below is an excerpt from my 
>> log files. For some reason the proxy server is not able to route the 
>> message to user2 therefore it sends a message to user1 changing the 
>> SIP header from 100 to 407(Proxy authorization needed).
>>
>>
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: 
>> DBG:tm:build_local: using FROM=> >;tag=232523
>>  >, TO=> > 
>> ;tag=9360cfebad69401b0dbd073e6bd6f131.188c
>>  >, CSEQ_N=
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: 
>> incoming reply
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]: 
>> DBG:core:parse_msg: SIP Request:
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: 
>> DBG:tm:t_should_relay_response: T_code=100, new_code=407
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]: 
>> DBG:core:parse_msg:  method:  
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: 
>> DBG:tm:t_pick_branch: picked branch 0, code 407 (prio=501)
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]: 
>> DBG:core:parse_msg:  uri: > >
>> Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: 
>> DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=407, 
>> flags=2
>>
>>
>> Could anyone give me some insights on how to go around this problem? 
>> Any help will be appreaciated.
>>
>> regards,
>> james
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Changing Opensips port number

2010-10-13 Thread Bogdan-Andrei Iancu

Hi James,

how do you set the port? via "port" param or via the "listen" param? are 
you sure you restart was effective?

Regards,
Bogdan

James Mbuthia wrote:
> Hi,
>
> Sorry for the newbie question. How do you change the port number in 
> opensips? I want to change it from 5060 to something else however when 
> I do so on opensips.cfg and restart opensips, Opensips doesn't seem to 
> listen on the port I have specified. Do I need to change the number 
> elsewhere?
>
> regards
> james
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] loose_route and $du

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Stefano,

yes, it is possible - if the request does not have any other Route hdr 
to indicate the next hope, routing based on RURI will be done, so the 
"loose_route" function will not set any $du (destination URI) - this du 
is actually a kind of outbound proxy used by opensips when it wants to 
send the request to other destination than RURI.

Such a case is when your opensips is the last hop before the end client.

Regards,
Bogdan

Stefano Sasso wrote:
> Hello,
>   sorry for the newbie stupid question.
>
> Is it normal that after a loose_route() the $du is set to null?
>
> this is my config:
> if (loose_route()) {
> if (is_method("BYE")) {
> setflag(1); # do accounting ...
> setflag(3); # ... even if the transaction 
> fails
> } else if (is_method("INVITE")) {
> # even if in most of the cases is
> useless, do RR for
> # re-INVITEs alos, as some buggy
> clients do change route set
> # during the dialog.
> record_route();
> }
> # route it out to whatever destination was set
> by loose_route()
> # in $du (destination URI).
> xlog("[LOOSE_ROUTE] from $si:$sp method $rm
> SET_DESTINATION $du");
> route(1);
> }
>
> in my log file I have:
> Oct 13 12:09:42 TDCVIRX09 /usr/local/sbin/opensips[17647]:
> [LOOSE_ROUTE] from 192.168.6.132:10002 method BYE SET_DESTINATION
> 
>
> thanks
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] error in opensips-cp

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Leon,

Have you created the ocp_admin_privileges table ? see the install 
instructions for OpenSIPS control panel on 
http://opensips-cp.sourceforge.net/

Regards,
Bogdan

Leon Li wrote:
>
> Hi ,
>
> I followed the instruction in “build telephony system with opensips 
> 1.6” to install opensips-cp. However, an error comes up when access 
> the link http://localhost/cp.
>
> Failed to issue query, error message: MDB2 Edrror: no such table.
>
> I did import the two tables for cdr and smonitor, and both appears in 
> mysql.
>
> Any ideas? I can generate a log file as instructed.
>
> Regards
>
> Leon
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] failed to decode uri

2010-10-13 Thread Bogdan-Andrei Iancu

Again, can you post a trace of the message causing the error?

Regards,
Bogdan

thrillerbee wrote:
> I was forced to remove the OpenSIPS b2bua as it seems it was causing 
> the origination gateway to choke on these 200 OKs.  Dialogs are no 
> longer hanging, but I still see errors on the origination gateway that 
> are concerning:
>
> /usr/local/sbin/opensips[4399]: ERROR:core:parse_from_header: bad from 
> header
> /usr/local/sbin/opensips[4399]: ERROR:rr:record_route: From parsing failed
> /usr/local/sbin/opensips[4402]: ERROR:core:parse_to: unexpected char 
> [#000] in status 6: <<"2345678910" 
>
> I can't find anything out of the ordinary in traces.  What should I be 
> looking for?
> Again, any help would be appreciated.
>
> Thanks.
>
> On Tue, Oct 12, 2010 at 9:15 PM, thrillerbee  > wrote:
>
> To add a little more detail, after doing some from uri
> normalization, this is eventually passed on to an OpenSIPS
> instance running b2bua for top hiding.  I'm assuming this error
> indicates that the header param that is used to rebuild the
> original from uri is different than expected, but I don't see any
> discernible differences in a trace between the headers in the 200
> OK (to the INVITE) and the provisional responses received right
> before it.  The provisional responses are proxied; the 200 OK is not.
>
> Any help would be most appreciated.
>
> Thanks.
>
>
> On Tue, Oct 12, 2010 at 6:13 PM, thrillerbee
> mailto:thriller...@gmail.com>> wrote:
>
> I have dialogs getting stuck because my origination gateway
> (OpenSIPS proxy) fails to proxy 200 OKs to some INVITEs.  In
> the logs, I see:
> /usr/local/sbin/opensips[30336]: ERROR:uac:decode_uri: invalid
> base64 string"AAUECAEIAwUHCQ /usr/local/sbin/opensips[30336]: ERROR:uac:restore_uri: failed
> to decode uri
> /usr/local/sbin/opensips[30337]: ERROR:uac:decode_uri: invalid
> base64 string"AAUECAEIAwUHCQ /usr/local/sbin/opensips[30337]: ERROR:uac:restore_uri: failed
> to decode uri
>
> I am doing some normalization of the from uri with
> uac_replace_from.  What could cause this?
>
> Thanks.
>
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] failed to decode uri

2010-10-13 Thread Bogdan-Andrei Iancu

Hi,

regarding the decode_uri issue : the uac module stores a cookie in the 
RR header, so at Route time it will be able to restore the original FROM 
/ TO hdr. I suspect that the one of the parties (caller or callee) is 
messing around with this RR parameter (vsf , vst), most probably with 
the case sensitiveness (switching from lower to upper case).

If you post a SIP trace on proxy, I can point you to the bogus party.

Regards,
Bogdan

thrillerbee wrote:
> To add a little more detail, after doing some from uri normalization, 
> this is eventually passed on to an OpenSIPS instance running b2bua for 
> top hiding.  I'm assuming this error indicates that the header param 
> that is used to rebuild the original from uri is different than 
> expected, but I don't see any discernible differences in a trace 
> between the headers in the 200 OK (to the INVITE) and the provisional 
> responses received right before it.  The provisional responses are 
> proxied; the 200 OK is not.
>
> Any help would be most appreciated.
>
> Thanks.
>
> On Tue, Oct 12, 2010 at 6:13 PM, thrillerbee  > wrote:
>
> I have dialogs getting stuck because my origination gateway
> (OpenSIPS proxy) fails to proxy 200 OKs to some INVITEs.  In the
> logs, I see:
> /usr/local/sbin/opensips[30336]: ERROR:uac:decode_uri: invalid
> base64 string"AAUECAEIAwUHCQ /usr/local/sbin/opensips[30336]: ERROR:uac:restore_uri: failed to
> decode uri
> /usr/local/sbin/opensips[30337]: ERROR:uac:decode_uri: invalid
> base64 string"AAUECAEIAwUHCQ /usr/local/sbin/opensips[30337]: ERROR:uac:restore_uri: failed to
> decode uri
>
> I am doing some normalization of the from uri with
> uac_replace_from.  What could cause this?
>
> Thanks.
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] regular expression matching

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Gabriel,

why don't you use in DR 2 different rule with different "strip" and 
"pri_prefix" like:

rule1:
matches:00593
strips: 5 digits
pri_prefix: "7424"

rule2:
matches:001 (or you can use default rule with empty matching prefix)
strips: 0 digits
pri_prefix: "7424"

The GWs used by each rule can be the same, of course.

Regards,
Bogdan

Gabriel Bermudez wrote:
> hi bodgan
>
> true, the problem is my voip provider, all calls must be delivered
> with a "pri prefix" at the begining, but local calls must be delivered
> without the E.164 prefix and I receive all my calls in E.164 so I must
> replace this prefix with another on, so for a local call.
>
> local call (593 E.164 prefix)
> 005938 -> 74248
> international call
> 0015678942356 -> 74240015678942356
>
> I have managed to check for the special prefix with this expression
>
> do_routing();
> $avp(s:pri_local) = $(avp(s:dr_attrs){param.value,pri_local});
> $avp(s:pri_local_re) = "sip:"+$avp(s:pri_local)+"*";
> xlog("check if $ru matches $avp(s:pri_local_re)");
> if(avp_check("$ruri", "fm/$avp(s:pri_local_re)")) {
>$avp(s:pri_replace) = $(avp(s:dr_attrs){param.value,pri_replace});
>xlog("$avp(s:pri_local) must be replaced by $avp(s:pri_replace)");
>$avp(s:new_uri) = $rU;
>$avp(s:local_len) = $(avp(s:pri_local){s.len});
>$avp(s:new_uri) = $avp(s:pri_replace) +
> $(avp(s:new_uri){s.substr,$avp(s:local_len),0});
>xlog("new uri $avp(s:new_uri)");
>$rU = $avp(s:new_uri);
> }
>
> works, but I would like to do the replacement less verbose, for
> performace reasons mainly.  Can it be done?
>
> Regards,
>
> 2010/10/11 Bogdan-Andrei Iancu :
>   
>> Hi Gabriel,
>>
>> when using regexp ops, you cannot use variables in the regexp definition.
>>
>> But your I miss your logic here - if you do routing based on prefixes
>> (do_routing() ), why do you need an extra regexp checking for that rule
>> ? if the Rule matches, you already know that RURI looks like
>> "sip:7424005938...@a.b.c.d"
>>
>> Regards,
>> Bogdan
>>
>> Gabriel Bermudez wrote:
>> 
>>> Hi everyone,
>>>
>>> I'm having trouble creating a regex in the opensips.cfg file.  This is
>>> the snipplet
>>>
>>> modparam("drouting", "attrs_avp", '$avp(s:dr_attrs)')
>>> ...
>>> do_routing();
>>> $avp(s:special) = $(avp(s:dr_attrs){param.value,special});
>>> xlog("attrs/$avp(s:dr_attrs) special/$avp(s:special) rU/$ru
>>> ^sip:$avp(s:special)[0-9]*@");
>>> if (uri =~ "^sip:$avp(s:special)[0-9]*@") {
>>> xlog("do something to the R-URI");
>>> }
>>>
>>> the $avp(s:dr_attrs) gets loaded with "special=742400593", so the
>>> avp(s:special) has the value "742400593" and the the $ru variable
>>> which I presume must have the same value of the uri variable has
>>> "sip:7424005938...@a.b.c.d", but for some reason the expression is
>>> not returning true.  What I'm doing wrong.
>>>
>>> Regards,
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>   
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 15 - 19 November 2010, Edison, New Jersey, USA
>> www.voice-system.ro
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Permissions Module bug in 1.6.3 ?

2010-10-13 Thread Bogdan-Andrei Iancu

Thanks Logan,

The patch was already uploaded on SVN trunk and stable.

Regards,
Bogdan

logan wrote:
> Bogdan, I've applied the patch you provided and initial testing has 
> been successful. I'll keep you posted on the rest of my tests. Thank 
> you for your attention to this!
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS swallows BYEs

2010-10-13 Thread Bogdan-Andrei Iancu

David,

In this case, instead of using the "expensive" solution of a b2bua, why 
don't you take care of properly re-write the contact. The question is 
why do you replace the received contact with the IP of your opensips ?

Regards,
Bogdan

David Santiago wrote:
> I'm not explicitly routing that BYE, but as Andrew Pogrebennyk pointed
> out, since I'm "rewriting the Contact header with OpenSIPS address it
> is expected that BYE won't go any further than OpenSIPs proxy" and
> therefore my only solutions seems to be the configuration of the b2bua
> modules. Isn't it?
>
> On Mon, Oct 11, 2010 at 6:35 PM, Bogdan-Andrei Iancu
>  wrote:
>   
>> Hi David,
>>
>> ok, so the BYE is sent to the wrong destination - how do you route BYE in
>> your script? do you use the standard "loose_route" block ?
>>
>> Regards,
>> Bogdan
>>
>> David Santiago wrote:
>> 
>>> As I said it is being forwarded to the jain slee server that
>>> originated it. The effect is the same as if it was replied.
>>>
>>> The flow is:
>>> step 1:   [JAIN SLEE SERVER]   ---BYE-->[OPENSIPS][VOIP
>>> PROVIDER]
>>> step 2:   [JAIN SLEE SERVER]   <---BYE--[OPENSIPS][VOIP
>>> PROVIDER]
>>>
>>> that BYE should be forwarded to the VOIP provider, instead.
>>>
>>> I'm having a look at how to configure the b2bua thing should this
>>> problem be fixed.
>>>
>>> Thanks for the follow up, Bogdan.
>>>
>>>
>>> David
>>>
>>> On Fri, Oct 8, 2010 at 1:38 PM, Bogdan-Andrei Iancu
>>>  wrote:
>>>
>>>   
 Hi David,

 my question is what happens with the BYE:
  - is it replied by opensips ?
  - is it forwarded to some whatever destination ?

 Regards,
 Bogdan

 David Santiago wrote:

 
> Bogdan, right now it's being forwarded again to the slee server who sent
> it, as I'm basically using the configuration provided in
> http://www.opensips.org/html/docs/modules/1.6.x/dispatcher.html
>
> On Wed, Oct 6, 2010 at 4:16 PM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>> wrote:
>
>   Hi David,
>
>   Is the BYE replied or forwarded to whatever destination ?
>
>   probably  your record routing is somehow broken. OpenSIPS may
>   misroute the BYE because the invalid route set - posting the 200
>   OK for INVITE and the BYE will help in investigating this.
>
>   Regards,
>   Bogdan
>
>   David Santiago wrote:
>
>   Hi all,
>
>   I have a running OpenSIPS installation that I'm using for
>   testing purposes.
>
>   The fact is that I'm forwarding requests from a voip provider
>   to a jain slee server and everything is working fine (INVITEs,
>   ACKs, RTP flow,...), except for the BYEs generated from the
>   server side. They reach the OpenSIPs proxy and are not
>   forwarded to the voip provider in order to finish the call.
>
>   I'm not sure if I have to manually setup a route for this to
>   happen, or if this behaviour is only available by using the
>   B2BUA approach in OpenSIPS.
>
>
>   Thanks a lot!
>
>   David
>
>   
-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] $avp in codec_exists

2010-10-13 Thread Bogdan-Andrei Iancu

Hi Dimitri,

I don;t think it will pay off - especially that you can do it from 
script level (multi checks in "if" statement)

Regards,
Bogdan

DM wrote:
> Hi Bogdan,
>
> It is hard to implement with multiple codecs?, eg. if I store it like 
> this:
> $avp(s:codlist)="PCMA|G723|tel";
>
> So not with just one codec.
>
> Thanks,
>
> Dimitri
>
>
> 2010/7/14 Bogdan-Andrei Iancu  >
>
> thanks!
>
> Jozsef CZOMPO wrote:
> > Hi bogdan
> >
> > Thanks for the reply. I'm opened a feature request ticket: #3029568
> >
> >   Czo
> >
> > On 2010.07.14., at 17:33, Bogdan-Andrei Iancu wrote:
> >
> >
> >> Hi Jozsef,
> >>
> >> open a feature request on the project tracker for this small add-on
> >> - it
> >> sounds useful and it is not difficult to add.
> >>
> >> Regards,
> >> bogdan
> >>
> >> Jozsef CZOMPO wrote:
> >>
> >>> Hi
> >>>
> >>> I need to specify in database table the codec order per sip
> clients.
> >>> So i'm created a table with some fields. But if use
> >>> codec_exists('PCMU') it return true, when i'm using
> >>> codec_exists($avp(s:codec)) i'm getting syntax error, and if i'm
> >>> using
> >>> codec_exists('$avp(s:codec)') i'm getting false. When i'm use xlog
> >>> ( "L_ERR", "$avp(s:codec)"); the result is PCMU, so the
> avp_db_query
> >>> is working fine.
> >>>
> >>> So, how can i use this?
> >>>
> >>> Thank
> >>> Czo
> >>>
> >>>
> >>> ___
> >>> Users mailing list
> >>> Users@lists.opensips.org 
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>
> >>>
> >> --
> >> Bogdan-Andrei Iancu
> >> OpenSIPS Bootcamp
> >> 20 - 24 September 2010, Frankfurt, Germany
> >> www.voice-system.ro 
> >>
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org 
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro 
>
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] DNS Failover test

2010-10-13 Thread Duane Larson

I could be wrong but does your phone register with the proxy and then the
proxy create a location record for each user?  That would be how opensips
would know how to route it.

On Oct 13, 2010 1:04 PM, "James Mbuthia"  wrote:

Hi guys,

Am a SIP and Opensips newbie and am trying to create a proof of concept on a
unique (I think) Opensips and SIPS implementation. Basically am developing a
web-based SIP Phone based on PHP and using Opensips. Because of the nature
of the application, users will have the same domain and just one proxy
server which will process invites and registrations.  Based on my
understanding of SIP and Opensip, I have developed a UAC that can send
invites and registrations to a proxy server. My problem is coming when I
have to route a call from one agent to another. According to RFC 3263 the
proxy server needs to determine the sip server for the person receiving the
call. Know if the proxy server of the person receiving the call and the
person initiating the call is the same and both the caller and callee are on
the same domain than how does Opensips route the message?

I've worked on this the last few days and below is an excerpt from my log
files. For some reason the proxy server is not able to route the message to
user2 therefore it sends a message to user1 changing the SIP header from 100
to 407(Proxy authorization needed).

Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:build_local: using
FROM=
>;tag=232523
 >, TO=>
;tag=9360cfebad69401b0dbd073e6bd6f131.188c
 >, CSEQ_N=
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: incoming
reply
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg: SIP Request:
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:t_should_relay_response: T_code=100, new_code=407
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg:  method:  
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:t_pick_branch: picked branch 0, code 407 (prio=501)
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg:  uri:

>
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=407, flags=2

Could anyone give me some insights on how to go around this problem? Any
help will be appreaciated.

regards,
james

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Core dump in svn head

2010-10-13 Thread Dave May

Over the past couple weeks I have been getting occasional segfaults just
prior to (or perhaps in the process of) a t_relay() in my failure_route.
Still haven't gotten to the bottom of the root cause in my config, but I
was able to find and fix the symptomatic code in the
pre_print_uac_request() function in t_fwd.c (based on Bogdan's previous
fix for a similar problem).

Backtrace:
#0  0xb763ba55 in memcpy () from /lib/tls/i686/cmov/libc.so.6
#1  0xb72ede1d in pre_print_uac_request (t=0x972ddf78, branch=1,
request=0x972cec64) at t_fwd.c:177
#2  0xb72eeac0 in add_uac (t=0x972ddf78, request=0x7fff,
uri=0xbf8a250c, next_hop=0xbf8a2514, path=0xb7315100, proxy=0x0) at
t_fwd.c:401
#3  0xb72efbc0 in t_forward_nonack (t=0x972ddf78, p_msg=0xb7314de0,
proxy=0x0) at t_fwd.c:660
#4  0xb72fbdbf in w_t_relay (p_msg=0xb7314de0, proxy=0x0, flags=) at tm.c:1116
#5  0x08057810 in do_action (a=0x819f924, msg=0xb7314de0) at
action.c:1155
#6  0x080557ff in run_action_list (a=0x819e7cc, msg=0xb7314de0) at
action.c:140
#7  0x0805981e in do_action (a=0x819fed4, msg=0xb7314de0) at
action.c:821
#8  0x080557ff in run_action_list (a=0x819e584, msg=0xb7314de0) at
action.c:140
#9  0x0805981e in do_action (a=0x81a0390, msg=0xb7314de0) at
action.c:821
#10 0x080557ff in run_action_list (a=0x819d588, msg=0xb7314de0) at
action.c:140
#11 0x0805aef1 in run_top_route (a=0x819d588, msg=0xb7314de0) at
action.c:120
#12 0xb7304c2c in t_should_relay_response (Trans=0x972ddf78,
new_code=408, branch=0, should_store=0xbf8a3094,
should_relay=0xbf8a3098, cancel_bitmap=0xbf8a30e8, reply=0x)
at t_reply.c:621
#13 0xb730683d in relay_reply (t=0x972ddf78, p_msg=0x, branch=0,
msg_status=408, cancel_bitmap=0xbf8a30e8) at t_reply.c:1133
#14 0xb72f21e5 in fake_reply (t=0x972ddf78, branch=0, code=408) at
timer.c:259
#15 0xb72f3dd7 in timer_routine (ticks=3285, attr=0x0) at timer.c:365
#16 0x080c12c0 in start_timer_processes () at timer.c:325
#17 0x0806e37e in main (argc=Cannot access memory at address 0x7fff
) at main.c:872

Here is the patch to fix:

Index: modules/tm/t_fwd.c
===
--- modules/tm/t_fwd.c  (revision 7259)
+++ modules/tm/t_fwd.c  (working copy)
@@ -166,7 +166,7 @@
-request->REQ_METHOD);
 
/* copy dst_uri into branch (after branch route possible updated
it) */
-   if (request->dst_uri.len) {
+   if ( request->dst_uri.s && request->dst_uri.len>0 ) {
t->uac[branch].duri.s =
shm_resize(t->uac[branch].duri.s,
request->dst_uri.len);
if (t->uac[branch].duri.s==NULL) {


After the code change and a recompile, the following lines appear in my
logs, whereas previously at this spot OpenSIPS crashed:

Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:core:parse_uri:
uri too short: <> (-1) 
Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:tm:uri2proxy:
bad_uri: (null) 
Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]:
ERROR:tm:t_forward_nonack: failure to add branches 
Oct 12 16:04:13 sipdev1 /usr/sbin/opensips[31245]: ERROR:tm:w_t_relay:
t_forward_nonack failed

Dave.

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] DNS Failover test

2010-10-13 Thread James Mbuthia

Hi guys,

Am a SIP and Opensips newbie and am trying to create a proof of concept on a
unique (I think) Opensips and SIPS implementation. Basically am developing a
web-based SIP Phone based on PHP and using Opensips. Because of the nature
of the application, users will have the same domain and just one proxy
server which will process invites and registrations.  Based on my
understanding of SIP and Opensip, I have developed a UAC that can send
invites and registrations to a proxy server. My problem is coming when I
have to route a call from one agent to another. According to RFC 3263 the
proxy server needs to determine the sip server for the person receiving the
call. Know if the proxy server of the person receiving the call and the
person initiating the call is the same and both the caller and callee are on
the same domain than how does Opensips route the message?

I've worked on this the last few days and below is an excerpt from my log
files. For some reason the proxy server is not able to route the message to
user2 therefore it sends a message to user1 changing the SIP header from 100
to 407(Proxy authorization needed).


Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:build_local: using
FROM=
>;tag=232523
 >, TO=>
;tag=9360cfebad69401b0dbd073e6bd6f131.188c
 >, CSEQ_N=
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]: incoming
reply
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg: SIP Request:
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:t_should_relay_response: T_code=100, new_code=407
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg:  method:  
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:t_pick_branch: picked branch 0, code 407 (prio=501)
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19405]:
DBG:core:parse_msg:  uri:

>
Oct 13 08:26:53 CL-T020-483CL /usr/local/sbin/opensips[19406]:
DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=407, flags=2


Could anyone give me some insights on how to go around this problem? Any
help will be appreaciated.

regards,
james
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Relaying custom header(s) in presence

2010-10-13 Thread Ovidiu Sas

Hello all,

The current implementation of the pua module allows sending custom
headers in PUBLISH requests.
I would like to have the ability to save those headers in the presence
server and relay them to presence subscribers via notifications
(NOTIFY requests).
This would require adding a new field to the presentity table to store
the extra header(s).
Each registered event must provide a list of headers to store (none
given, none stored).

When a PUBLISH is received, if there are header(s) to be saved, those
header(s) will be saved along with the body.
When the NOTIFY request is prepared, if extra header(s) are available,
then those header(s) will be appended to the request.


Regards,
Ovidiu Sas

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] failed to decode uri

2010-10-13 Thread thrillerbee

I was forced to remove the OpenSIPS b2bua as it seems it was causing the
origination gateway to choke on these 200 OKs.  Dialogs are no longer
hanging, but I still see errors on the origination gateway that are
concerning:

/usr/local/sbin/opensips[4399]: ERROR:core:parse_from_header: bad from
header
/usr/local/sbin/opensips[4399]: ERROR:rr:record_route: From parsing failed
/usr/local/sbin/opensips[4402]: ERROR:core:parse_to: unexpected char [#000]
in status 6: <<"2345678910" wrote:

> To add a little more detail, after doing some from uri normalization, this
> is eventually passed on to an OpenSIPS instance running b2bua for top
> hiding.  I'm assuming this error indicates that the header param that is
> used to rebuild the original from uri is different than expected, but I
> don't see any discernible differences in a trace between the headers in the
> 200 OK (to the INVITE) and the provisional responses received right before
> it.  The provisional responses are proxied; the 200 OK is not.
>
> Any help would be most appreciated.
>
> Thanks.
>
>
> On Tue, Oct 12, 2010 at 6:13 PM, thrillerbee wrote:
>
>> I have dialogs getting stuck because my origination gateway (OpenSIPS
>> proxy) fails to proxy 200 OKs to some INVITEs.  In the logs, I see:
>> /usr/local/sbin/opensips[30336]: ERROR:uac:decode_uri: invalid base64
>> string"AAUECAEIAwUHCQ> /usr/local/sbin/opensips[30336]: ERROR:uac:restore_uri: failed to decode
>> uri
>> /usr/local/sbin/opensips[30337]: ERROR:uac:decode_uri: invalid base64
>> string"AAUECAEIAwUHCQ> /usr/local/sbin/opensips[30337]: ERROR:uac:restore_uri: failed to decode
>> uri
>>
>> I am doing some normalization of the from uri with uac_replace_from.  What
>> could cause this?
>>
>>  Thanks.
>>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

2010-10-13 Thread Richard Revels

I'm not sure that process persistence is what Brett was looking for but rather 
Dialog persistence.  I have found that local memcache support is very fast and 
takes care of this type of need quite well.  

Using a unique key, made up perhaps of the SIP call-id and type of value like 
"B2Bwhatever_callerid" and the value you desire it to contain will allow you to 
store, retrieve, and delete the value at will from any point in call 
processing.  Plus you can pass in an expire time so it will be deleted in the 
event something goes wrong with the call.  If you want to get real fancy you 
could even set up a memcached server and do those same functions from 
distributed Proxies.

http://www.opensips.org/Resources/DocsTutMemcache.

There are always a thousand ways to do any one thing in opensips, like storing 
the avp to the database 

case "INVITE" :
avp_db_store("$ci", "$avp(s:epochtime)"); 
#epochtime set in "set variables route" - store it to db with call id as uuid.


and grabbing it back

case "BYE" :
avp_db_load("$ci", "a"); #get all records from db 
stored with this call id as uuid.
#this caused epochtime avp to become multi.
avp_db_delete("$ci", "a"); #get rid of db entries for 
this call


(this example doesn't exactly match your need but you get the idea) but 
memcache is one way to do this job.


Richard

On Oct 13, 2010, at 3:13 AM, Anca Vamanu wrote:

> Hi Brett,
> 
> I think using script variables should work - 
> http://www.opensips.org/Resources/DocsCoreVar#varscript.
> 
> Regards,
> Anca
> 
> On 10/13/2010 01:31 AM, Brett Woollum wrote:
>> 
>> Hi Anca,
>> 
>> I figured out the solution that you were referring to. I added the 
>> local_route section and changed the header in there, and it works.
>> 
>> The only problem now is that the AVP variable I set in route doesn't seem to 
>> be persistent into the new request inside local_route. This is a problem 
>> because I need some way to send the variables to local_route so that I can 
>> put them into the new request. Is there a way to make an AVP variable 
>> persistent into the local_route as well? Or maybe there is a better way to 
>> make these variables accessible from inside local_route?
>> 
>> Here is my working code (with statically set caller id. I need to make this 
>> dynamic):
>> local_route {
>> if (is_method("INVITE")) {
>> remove_hf("From");
>> append_hf("From: \"TEST3\" ;tag=$ft\r\n");
>> #exit;
>> }
>> if (is_method("BYE") ) {
>> acc_log_request("internally generated BYE");
>> }
>> }
>> 
>> Brett Woollum
>> br...@woollum.com
>> 
>> 
>> - Original Message -
>> From: "Anca Vamanu" 
>> To: users@lists.opensips.org
>> Sent: Tuesday, October 12, 2010 2:37:55 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question
>> 
>> Hi Brett,
>> 
>> The b2bua generates a new request with the info from the received one. The 
>> problem is with the changes upon a request not being visible to the 
>> following functions called from the script. 
>> So you have to change the header on the generated one, in local route.
>> 
>> Regards,
>> Anca
>> 
>> 
>> On 10/12/2010 10:24 AM, Brett Woollum wrote:
>> In addition, the uac_replace_from(display,uri); function does not seem to 
>> work either. I tested this function by putting:
>> 
>> uac_replace_from("TEST", "sip:12...@1.2.3.4");
>> 
>> immediately above the call to B2B_INIT_REQUEST and the outbound "From:" 
>> header remains unchanged (incorrect).
>> 
>> Brett Woollum
>> br...@woollum.com
>> 
>> 
>> - Original Message -
>> From: "Brett Woollum" 
>> To: "users-request" 
>> Sent: Monday, October 11, 2010 11:53:39 PM GMT -08:00 US/Canada Pacific
>> Subject: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question
>> 
>> Hello,
>> 
>> I have a working implementation of the B2BUA modules for OpenSIPS. One 
>> problem I am having is that the B2BUA module doesn't seem to respect the 
>> "From:" header that I updated before B2B_INIT was called (at the end of the 
>> script). Before I call B2B_INIT, my script looks up the correct caller-id 
>> from a database and replaces the "From:" header with the new information. 
>> Here is that code:
>> 
>> xlog("This is an external call...");
>> avp_db_query("select id from subscriber where username=\"$fU\" AND 
>> domain=\"1.2.3.4\"", "$avp(s:subscriber_id)");
>> if(is_avp_set("$avp(s:subscriber_id)")){  
>> avp_db_query("select 
>> external_cid_name,external_cid_uri,external_cid_uri_domain from cid where 
>> subscriber_id=\"$avp(s:subscriber_id)\"", 
>> "$avp(s:external_cid_name);$avp(s:external_cid_uri);$avp(s:external_cid_uri_domain)");
>> remove_hf("From");
>> append_hf("From:

[OpenSIPS-Users] Changing Opensips port number

2010-10-13 Thread James Mbuthia

Hi,

Sorry for the newbie question. How do you change the port number in
opensips? I want to change it from 5060 to something else however when I do
so on opensips.cfg and restart opensips, Opensips doesn't seem to listen on
the port I have specified. Do I need to change the number elsewhere?

regards
james
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] loose_route and $du

2010-10-13 Thread Stefano Sasso

Hello,
  sorry for the newbie stupid question.

Is it normal that after a loose_route() the $du is set to null?

this is my config:
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
} else if (is_method("INVITE")) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy
clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set
by loose_route()
# in $du (destination URI).
xlog("[LOOSE_ROUTE] from $si:$sp method $rm
SET_DESTINATION $du");
route(1);
}

in my log file I have:
Oct 13 12:09:42 TDCVIRX09 /usr/local/sbin/opensips[17647]:
[LOOSE_ROUTE] from 192.168.6.132:10002 method BYE SET_DESTINATION


thanks

-- 
Stefano Sasso
http://stefano.dscnet.org/

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Using b2bua top hiding plus dispatcher module

2010-10-13 Thread David Santiago

I'll check it later, but the fact is that I have to solve first the
problem that prevents the INVITEs generated by OpenSIPs to be sent to
the internal jain slee server.

I have confirmed that after the call to ds_select(), the value of $du
points to such server (which has been obtained from the
/opt/opensipsnotls/etc/opensips/dispatcher.list file), but the INVITEs
are not sent there, but to the public IP address of the OpenSIPs
server. Without the b2bua "setup", the dispatcher has been working
fine in other tests I've made...


Regards,
David

On Tue, Oct 12, 2010 at 11:22 AM, Anca Vamanu  wrote:
> Hi David,
>
> The only thing that won't work as you like is the append_hf - it won't
> work as you like since the b2bua module won't see that header. You need
> to do that for the new request generated by the b2bua and put that code
> in the local_route.
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>
> On 10/11/2010 01:48 PM, David Santiago wrote:
>> Hi all,
>>
>> After cleaning up the initialization problems in the b2bua related
>> modules I'm testing a configuration I want to use in order to enable
>> the "top hiding" scenario, plus using the dispatcher module so I can
>> configure the internal jain slee server that receives the requests
>> from the opensips proxy, like this:
>>
>> VOIP PROVIDER<--->    OPENSIPS<--->  INTERNAL JAIN SLEE SERVER
>>
>> I will write here the configuration I'm using, taken from the
>> configuration example available, that results in the INVITEs received
>> from the VOIP PROVIDER being sent by OPENSIPS to itself (to its public
>> address, in particular), ad infinitum.
>>
>> The modparam("dispatcher", "list_file",
>> "/opt/opensipsnotls/etc/opensips/dispatcher.list") entry points to the
>> file required by the dispatcher module to determine the ip and port
>> where the internal jain slee server is running. Furthermore, our
>> service implementation requires the addition of the
>> P-Preferred-Service header, which is done with append_hf().
>>
>> Any hint would be appreciated. Thanks in advance!
>>
>>
>>
>> # Debugging mode:
>> debug=9
>> memlog=6
>> fork=yes
>> auto_aliases=no
>> log_stderror=no
>>
>> disable_dns_blacklist=yes
>>
>> listen=10.122.54.163:5060 # private opensips ip
>> # alias=OSIPS_REALM
>> #port=OSIPS_PORT
>>
>>
>> sip_warning=0
>> #server_header="SRV_SIGNATURE"
>> #user_agent_header="PP_USER_AGENT"
>>
>> #mhomed=yes
>> #memlog=2
>> check_via=no
>> dns=off
>> rev_dns=off
>> children=8
>> disable_tcp=yes
>> log_facility=LOG_LOCAL0
>> # for more info: opensips -h
>>
>> # -- module loading --
>>
>> mpath="/opt/opensipsnotls/lib/opensips/modules/"
>>
>> #loadmodule "db_mysql.so"
>> #loadmodule "xlog.so"
>> loadmodule "textops.so"
>> loadmodule "maxfwd.so"
>> loadmodule "rr.so"
>> loadmodule "sl.so"
>> loadmodule "tm.so"
>> loadmodule "signaling.so"
>> loadmodule "b2b_entities.so"
>> loadmodule "b2b_logic.so"
>> loadmodule "usrloc.so"
>> loadmodule "registrar.so"
>> loadmodule "mi_fifo.so"
>> loadmodule "dispatcher.so"
>>
>> # - setting module-specific parameters ---
>> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>>
>>
>> #modparam("usrloc", "db_url", 
>> "mysql://opensips:opensip...@localhost/opensips")
>> #modparam("usrloc", "db_mode", 2)
>>
>> modparam("tm", "pass_provisional_replies", 1)
>>
>> modparam("dispatcher", "list_file",
>> "/opt/opensipsnotls/etc/opensips/dispatcher.list")
>>
>> modparam("b2b_entities", "server_address",
>> "sip:MY_OPENSIPS_PUBLIC_IP_GOES_HERE:5060")
>>
>>
>> #modparam("b2b_logic", "script_scenario",
>> "/home/anca/work/opensips/modules/b2b_logic/scenario_script.xml")
>> #modparam("b2b_logic", "extern_scenario",
>> "/home/anca/work/opensips/modules/b2b_logic/scenario_extern.xml")
>> modparam("b2b_entities", "script_req_route", "b2b_request")
>> modparam("b2b_entities", "script_reply_route", "b2b_reply")
>>
>> route[b2b_request] {
>>    xlog("b2b_request ($ci)\n");
>> }
>>
>>
>> route[b2b_reply] {
>>    xlog("b2b_reply ($ci)\n");
>> }
>>
>>
>> route {
>>
>>    xlog("mozaa - Here we go...");
>>
>>    if (!mf_process_maxfwd_header("10")) {
>>      sl_send_reply("483","Too Many Hops");
>>      exit;
>>    };
>>
>>    if (msg:len>= 2380 ) {
>>      sl_send_reply("513", "Message too big");
>>      exit;
>>    };
>>
>>    if (!method=="REGISTER")
>>      record_route();
>>
>>    # subsequent messages withing a dialog should take the
>>    # path determined by record-routing
>>    if (loose_route()) {
>>      # mark routing logic in request
>>      append_hf("P-hint: rr-enforced\r\n");
>>      route(1);
>>    };
>>
>>    if(is_method("INVITE")&&    /* only for Invite messages from bob*/
>>            !(src_ip == "10.122.54.163"&&  src_port ==5060)) /* skip
>> Invite messages generated by the server*/
>>    {
>>      # DO NOT call t_newtran() on this request ->  it will result in the
>> transaction never being deleted
>>      ##b2b_init_request(

Re: [OpenSIPS-Users] OpenSIPS swallows BYEs

2010-10-13 Thread David Santiago

I'm not explicitly routing that BYE, but as Andrew Pogrebennyk pointed
out, since I'm "rewriting the Contact header with OpenSIPS address it
is expected that BYE won't go any further than OpenSIPs proxy" and
therefore my only solutions seems to be the configuration of the b2bua
modules. Isn't it?

On Mon, Oct 11, 2010 at 6:35 PM, Bogdan-Andrei Iancu
 wrote:
> Hi David,
>
> ok, so the BYE is sent to the wrong destination - how do you route BYE in
> your script? do you use the standard "loose_route" block ?
>
> Regards,
> Bogdan
>
> David Santiago wrote:
>>
>> As I said it is being forwarded to the jain slee server that
>> originated it. The effect is the same as if it was replied.
>>
>> The flow is:
>> step 1:   [JAIN SLEE SERVER]   ---BYE-->    [OPENSIPS]        [VOIP
>> PROVIDER]
>> step 2:   [JAIN SLEE SERVER]   <---BYE--    [OPENSIPS]        [VOIP
>> PROVIDER]
>>
>> that BYE should be forwarded to the VOIP provider, instead.
>>
>> I'm having a look at how to configure the b2bua thing should this
>> problem be fixed.
>>
>> Thanks for the follow up, Bogdan.
>>
>>
>> David
>>
>> On Fri, Oct 8, 2010 at 1:38 PM, Bogdan-Andrei Iancu
>>  wrote:
>>
>>>
>>> Hi David,
>>>
>>> my question is what happens with the BYE:
>>>  - is it replied by opensips ?
>>>  - is it forwarded to some whatever destination ?
>>>
>>> Regards,
>>> Bogdan
>>>
>>> David Santiago wrote:
>>>

 Bogdan, right now it's being forwarded again to the slee server who sent
 it, as I'm basically using the configuration provided in
 http://www.opensips.org/html/docs/modules/1.6.x/dispatcher.html

 On Wed, Oct 6, 2010 at 4:16 PM, Bogdan-Andrei Iancu
 mailto:bog...@voice-system.ro>> wrote:

   Hi David,

   Is the BYE replied or forwarded to whatever destination ?

   probably  your record routing is somehow broken. OpenSIPS may
   misroute the BYE because the invalid route set - posting the 200
   OK for INVITE and the BYE will help in investigating this.

   Regards,
   Bogdan

   David Santiago wrote:

       Hi all,

       I have a running OpenSIPS installation that I'm using for
       testing purposes.

       The fact is that I'm forwarding requests from a voip provider
       to a jain slee server and everything is working fine (INVITEs,
       ACKs, RTP flow,...), except for the BYEs generated from the
       server side. They reach the OpenSIPs proxy and are not
       forwarded to the voip provider in order to finish the call.

       I'm not sure if I have to manually setup a route for this to
       happen, or if this behaviour is only available by using the
       B2BUA approach in OpenSIPS.

       Thanks a lot!

       David

       ___
       Users mailing list
       Users@lists.opensips.org 
       http://lists.opensips.org/cgi-bin/mailman/listinfo/users

   --     Bogdan-Andrei Iancu
   OpenSIPS Bootcamp
   15 - 19 November 2010, Edison, New Jersey, USA
   www.voice-system.ro 

 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

>>>
>>> --
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Bootcamp
>>> 15 - 19 November 2010, Edison, New Jersey, USA
>>> www.voice-system.ro
>>>
>>>
>>>
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] About Nat Traversal and Stun

2010-10-13 Thread Paul Smith

Hi,
stun module allows you to run a stun server on the opensips platform... 
so your device sith the stun client uses your proxy ip(s) to access a 
stun server.
Paul

CheeWii wrote:
> Hi,
>  
>I want to solve the nat traversal problem, and I read the great 
> book "Building Telephony System with Opensips 1.6.0". Now I encounter 
> some doubts.
>   
>From the book,I know the STUN solution is considered a near-end NAT 
> traversal solution. And I think a near-end NAT traversal solution is 
> implemented on the client-side. Why does Opensips have the module 
> named stun? What is the relationship of the stun module in Opensips 
> and the stun in client?  
>  
>   In addition,what is  the main and effective solutions in Opensips to 
> solve the nat traversal problem?
>  
>   Best wishes~
>  
> CheeWii
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

2010-10-13 Thread Anca Vamanu


Hi Brett,

I think using script variables should work - 
http://www.opensips.org/Resources/DocsCoreVar#varscript.


Regards,
Anca

On 10/13/2010 01:31 AM, Brett Woollum wrote:

Hi Anca,

I figured out the solution that you were referring to. I added the 
local_route section and changed the header in there, and it works.


The only problem now is that the AVP variable I set in route doesn't 
seem to be persistent into the new request inside local_route. This is 
a problem because I need some way to send the variables to local_route 
so that I can put them into the new request. Is there a way to make an 
AVP variable persistent into the local_route as well? Or maybe there 
is a better way to make these variables accessible from inside 
local_route?


Here is my working code (with statically set caller id. I need to make 
this dynamic):

local_route {
if (is_method("INVITE")) {
remove_hf("From");
append_hf("From: \"TEST3\" ;tag=$ft\r\n");
#exit;
}
if (is_method("BYE") ) {
acc_log_request("internally generated BYE");
}
}

Brett Woollum
br...@woollum.com


- Original Message -
From: "Anca Vamanu" 
To: users@lists.opensips.org
Sent: Tuesday, October 12, 2010 2:37:55 AM GMT -08:00 US/Canada Pacific
Subject: Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

Hi Brett,

The b2bua generates a new request with the info from the received one. 
The problem is with the changes upon a request not being visible to 
the following functions called from the script.

So you have to change the header on the generated one, in local route.

Regards,
Anca


On 10/12/2010 10:24 AM, Brett Woollum wrote:

In addition, the uac_replace_from(display,uri); function does not
seem to work either. I tested this function by putting:

uac_replace_from("TEST", "sip:12...@1.2.3.4");

immediately above the call to B2B_INIT_REQUEST and the outbound
"From:" header remains unchanged (incorrect).

Brett Woollum
br...@woollum.com


- Original Message -
From: "Brett Woollum" 
To: "users-request" 
Sent: Monday, October 11, 2010 11:53:39 PM GMT -08:00 US/Canada
Pacific
Subject: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

Hello,

I have a working implementation of the B2BUA modules for OpenSIPS.
One problem I am having is that the B2BUA module doesn't seem to
respect the "From:" header that I updated before B2B_INIT was
called (at the end of the script). Before I call B2B_INIT, my
script looks up the correct caller-id from a database and replaces
the "From:" header with the new information. Here is that code:

xlog("This is an external call...");
avp_db_query("select id from subscriber where username=\"$fU\" AND
domain=\"1.2.3.4\"", "$avp(s:subscriber_id)");
if(is_avp_set("$avp(s:subscriber_id)")){
avp_db_query("select
external_cid_name,external_cid_uri,external_cid_uri_domain from
cid where subscriber_id=\"$avp(s:subscriber_id)\"",

"$avp(s:external_cid_name);$avp(s:external_cid_uri);$avp(s:external_cid_uri_domain)");
remove_hf("From");
append_hf("From:
\"$avp(s:external_cid_name)\"

;tag=$ft\r\n");
}

The resulting "From": header is perfect (it includes the new
caller-id information) and works without problems when I don't use
the B2BUA module. It seems as though the B2BUA module (which is
called last in the script) does not respect the new header, and
instead processes the call using whatever was in $fu when the
script started. Unfortunately $fu is not writable, so I can't
directly change it.

What are some possible work-arounds to get the B2BUA module to
create the new call but include the "From:" header that I've
updated earlier in the script?

Any ideas are greatly appreciated!

Brett Woollum
br...@woollum.com


___ Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
   




--
Anca Vamanu
www.voice-system.ro
   

___ Users mailing list 
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
   



--
Anca Vamanu
www.voice-system.ro

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] How opensis can manage different ports.

[OpenSIPS-Users] [OpenSIPS LiveDVD]Register OK but call failed:prompt "420 invalid destination"

[OpenSIPS-Users] (no subject)

[OpenSIPS-Users] Static Routes in OpenSIPS proxy

[OpenSIPS-Users] Static Routes in OpenSIPS proxy

Re: [OpenSIPS-Users] Core dump in svn head

Re: [OpenSIPS-Users] Changing Opensips port number

Re: [OpenSIPS-Users] DNS Failover test

[OpenSIPS-Users] Services management - question about proper module

Re: [OpenSIPS-Users] Serialforking failure, with lcr:parse_phostport: too many colons in udp:: 0

Re: [OpenSIPS-Users] B2BUA Transfers Intermittently Working/Not Working (Bug?)

Re: [OpenSIPS-Users] Dialplan problem ?

[OpenSIPS-Users] B2BUA Transfers Intermittently Working/Not Working (Bug?)

Re: [OpenSIPS-Users] Core dump in svn head

Re: [OpenSIPS-Users] DNS Failover test

Re: [OpenSIPS-Users] Changing Opensips port number

Re: [OpenSIPS-Users] loose_route and $du

Re: [OpenSIPS-Users] error in opensips-cp

Re: [OpenSIPS-Users] failed to decode uri

Re: [OpenSIPS-Users] failed to decode uri

Re: [OpenSIPS-Users] regular expression matching

Re: [OpenSIPS-Users] Permissions Module bug in 1.6.3 ?

Re: [OpenSIPS-Users] OpenSIPS swallows BYEs

Re: [OpenSIPS-Users] $avp in codec_exists

Re: [OpenSIPS-Users] DNS Failover test

[OpenSIPS-Users] Core dump in svn head

[OpenSIPS-Users] DNS Failover test

[OpenSIPS-Users] Relaying custom header(s) in presence

Re: [OpenSIPS-Users] failed to decode uri

Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

[OpenSIPS-Users] Changing Opensips port number

[OpenSIPS-Users] loose_route and $du

Re: [OpenSIPS-Users] Using b2bua top hiding plus dispatcher module

Re: [OpenSIPS-Users] OpenSIPS swallows BYEs

Re: [OpenSIPS-Users] About Nat Traversal and Stun

Re: [OpenSIPS-Users] B2BUA "From" Header & Caller-ID Question

36 matches

Site Navigation

Mail list logo

Footer information