[OpenSIPS-Users] OpenSIPS with FS as load balancer

[Abid Saleem]

Hi All,
I am new to OpenSIPS. We are trying to use OpenSIPS with FreeSWITCH for load 
balancing and High Availability. We plan to have 2 OpenSIPS servers as load 
balancers in Active-Standby mode and 4 FS servers for handling RTP/media and 
other functions. 
I have two confusions.
1- Who will act as a Registrar in this kind of setup, OpenSIPS or FS. What are 
the pros and cons with both options.2- Who will perform B2BUA function OpenSIPS 
or FS?3- Is B2BUA function required in Class 4 traffic exchange only scenario?
Please help me answer these questions if someone can. Appreciate your time.
Regards--Abid SaleemTechnical Manager NGNTerminus Technologies  
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Re: [OpenSIPS-Users] Asterisk Cluster Scenario

[Duane Larson]

I would say "Yes.  Very possible.".  It will involve a good deal of reading,
studying and testing, but i am 99% sure what you want to do has been done
before.

On Dec 1, 2010 1:02 AM, "jameszhou"  wrote:

Hi:
i want to use opensips and asterisk to server a callcenter about 2000
agents.Does it possible and stable?


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Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[ha do]

Hi Tijmen

it works fine right now for callcontrol hangups a call in my last email
can i request to make the instruction on config.sample.ini of the OPENSIPS part
callcontrol uses the mi_datagram to make it clearer not confusing with mi_fifo
and the socket_path is the same socket_name in mi_datagram config of opensips
it should be good for someone, likes me

i still need help
> 2 opensips2 is a gateway
>endpoint-opensips1---opensips2internet-VOIP service 
provider
So basically opensips2 is just a gateway to opensips1?
[ha] yes,how to config cdrtool rating calls for opensips2


Thank you
Ha


- Original Message 
From: Tijmen de Mes 
To: OpenSIPS users mailling list 
Sent: Tue, November 30, 2010 3:44:12 PM
Subject: Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

Hi,

On Nov 25, 2010, at 10:06 AM, ha do wrote:
> and opensips.cfg with fifo config
> # - mi_fifo params -
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0600)
> modparam("mi_fifo", "fifo_group", 0)
> modparam("mi_fifo", "fifo_group", "root")
> modparam("mi_fifo", "fifo_user", 0)
> modparam("mi_fifo", "fifo_user", "root")
> 
> and the callcontrol config.ini
> [CallControl]
> group   = root
> [OpenSIPS]
> socket_path = /tmp
> ;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i need 100
> terminated i need to increase to 100, right???
> 
> how to fix the issue
Try  using socket_path=/tmp/socket in config.ini. The socket path should match 
the "mi_datagram" ,"socket_name"
in the opensips config.

Tijmen de Mes
AG Projects


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Re: [OpenSIPS-Users] Help with opensips 1.63 for newbie

[Bogdan-Andrei Iancu]

If you enabled to multidomain support, add in domain table the IP or 
domain that is served by your opensips. And do a restart.


Regards,
Bogdan

PS: keep the list at CC all the time!

paul.gor...@gmail.com wrote:

I uncomented all relevant sections in the config to enable mysql auth and 
multi-domain support. From debug log I see that opensips tries to retransmit 
some stuff, I guess this may be the cause for the loop.
I use nated box with no firewall, can this cause the retransmissions? Is there 
a sample config for nated opensips?


-Original Message-
From: Bogdan-Andrei Iancu
Sent:  11/30/2010 3:18:55 PM
Subject:  Re: [OpenSIPS-Users] Help with opensips 1.63 for newbie

Hi Paul,

The "too many hops" reply means your REGISTER request loops on opensips 
- it loops as opensips fails (in the script logic) to see that the 
request must be locally handled and not forwarded.


Have you enabled the multi-domain support in the default script? or it 
is the unchanged script that comes with opensips ?  I'm asking just to 
guide you on troubleshooting this.


Regards,
Bogdan

paul.gor...@gmail.com wrote:
  

Hi,
i installed opensips under suse with mysql and multi-domain support, then added 2 users. Unfortunately I cannot get these users registered, keep getting "too many hops" message. I use standard opensips config which came with the software.  
Anybody can hint me on what can be wrong,please?  


Paul.

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Re: [OpenSIPS-Users] MySQL LAST_INSERT_ID() safe?

[Bogdan-Andrei Iancu]

Hi Chris,

It is safe, as opensips has a mysql connection for each process and a 
script route is continuously executed by a single process.  So your 
queries will go via same connection with no risk to have something else 
between.


Regards,
Bogdan

Chris Maciejewski wrote:

Hi,

I am trying to figure out if is it safe to use in OpenSIPs config file:

avp_db_query("INSERT INTO some_table VALUES ('data1')");
avp_db_query("SELECT LAST_INSERT_ID()","$avp(last_id);");

// do stuff with $avp(last_id);

How does OpenSIPs connects to MySQL sever? Does it use some smart
connection pooling maybe, which could result in unexpected behaviour?

Regards,
Chris

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Re: [OpenSIPS-Users] Help with opensips 1.63 for newbie

[Bogdan-Andrei Iancu]

Hi Paul,

The "too many hops" reply means your REGISTER request loops on opensips 
- it loops as opensips fails (in the script logic) to see that the 
request must be locally handled and not forwarded.


Have you enabled the multi-domain support in the default script? or it 
is the unchanged script that comes with opensips ?  I'm asking just to 
guide you on troubleshooting this.


Regards,
Bogdan

paul.gor...@gmail.com wrote:

Hi,
i installed opensips under suse with mysql and multi-domain support, then added 2 users. Unfortunately I cannot get these users registered, keep getting "too many hops" message. I use standard opensips config which came with the software.  
Anybody can hint me on what can be wrong,please?  


Paul.

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Re: [OpenSIPS-Users] BYE processing and AVP

[Bogdan-Andrei Iancu]

Hi Anton,

if you are looking for dialog- persistent variables, use the dialog 
module and the dialog variables:

  http://www.opensips.org/html/docs/modules/1.6.x/dialog.html
  http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id295169

Note that you need to create the dialog (at INVITE time) via 
"create_dialog()" function before start using the dialog variables.


Regards,
Bogdan

Anton Zagorskiy wrote:


Ok, thanks!

 

 

 

 


2




 

 

*From:* users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *mayamatakeshi

*Sent:* Tuesday, November 30, 2010 5:29 PM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] BYE processing and AVP

 

 

On Tue, Nov 30, 2010 at 11:19 PM, mayamatakeshi 
mailto:mayamatake...@gmail.com>> wrote:


 

2010/11/30 Anton Zagorskiy >


 


Why BYE isn't a part of the transaction?


From RFC3261:

Specifically, a SIP transaction consists of a single request and any responses 
to
   that request, which include zero or more provisional responses and
   one or more final responses.

But when opensips docs say transaction, it is also implying that you 
are using module tm (if you don't use tm functions, then you are doing 
stateless processing and no AVP will be saved). 



For completeness,
the BYE is part of the dialog established by the INVITE transaction 
that ended when "200 OK" was received. But the BYE itself starts 
another transaction.
So, the AVPs you set in the processing of the INVITE will be available 
(as long as you use module tm) when your process responses for that 
INVITE (180, 183, 200 etc), but not when you are handling other 
requests in the dialog.
 

 



WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593

a.zagors...@oyster-telecom.ru


www.oyster-telecom.ru 


From: users-boun...@lists.opensips.org

[mailto:users-boun...@lists.opensips.org
] On Behalf Of
mayamatakeshi
Sent: Tuesday, November 30, 2010 4:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP



On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
mailto:a.zagors...@oyster-telecom.ru>> wrote:
Hello.

During INVITE processing I'm setting up some avp variables.
When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:

if (has_totag())
{
   if (loose_route())
   {
   if (is_method("BYE"))
   {
   xlog("*** BYE $avp(i:2)");

From the docs
(http://www.opensips.org/Resources/DocsCoreVar#toc2):

"AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present. "

br,
takeshi


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Re: [OpenSIPS-Users] Debugging variables in failure_route

[Bogdan-Andrei Iancu]

Hi Bill,

you can use t_local_replied("last") in failure route to see if the 408 
was locally generated or was received. See:

   http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id294010

Regards,
bogdan

Bill W. wrote:

Hello everyone,

I'm trying to debug an issue where something is intermittently sending a
"408 Request timed out" which is being caught by t_check_status("408")
in a failure_route.

And actually, I think opensips might be generating it internally because
the INVITE message id is very different than the message id of the 408.
 ($mi=385 vs $mi=1)

Regardless, all the variables I've tried to log inside the failure_route
pertain to the initial INVITE, so they are of no help.  I even tried
$(rs) and $(rr) but those both show .

So the question is: are there any variables I can log within the
failure_route to determine the source of that SIP 408?

Thanks!
Bill W.



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Re: [OpenSIPS-Users] Trouble parsing VSA in Radius Reply (auth)

[Bogdan-Andrei Iancu]

Hi Brett,

about VSA, see fix 7410 (1.6 branch) / 7439 (trunk).It is related to 
handling the radius AVP that have a vendor id.


Regards,
Bogdan

Brett Nemeroff wrote:

Hello All,
I'm trying to parse a VSA reply from a radius_send_auth call. My 
second set looks like this:
modparam("aaa_radius","sets","ipauthreply  =  (  h323-currency = 
$avp(s:h323-currency), h323-credit-amount = $avp(s:h323-credit-amount) 
) ")


Opensips starts up, which suggests to me that it's finding 
these attributes in the dictionary just fine. I've tested that by 
changing the attribute to something like h323-currency2 and it fails 
(as expected). The send radius packet looks great and I've determined 
with a pcap that the reply is also properly formated. The reply has 
the two above attributes with the vendor id of 9 (cisco). Those two 
attributes do in fact show up in my dictionary as cisco attribtues:


VENDOR  Cisco   9
...
ATTRIBUTE   h323-credit-amount  101 string 
 Cisco
ATTRIBUTE   h323-currency   110 string 
 Cisco

...

When I get a reply, this shows up in the error log (once for each 
attribute I'm looking for:
"Nov 16 12:01:22 srv1158 /usr/local/sbin/opensips[25799]: 
ERROR:aaa_radius:send_auth_func: attribute was not found in received 
radius message "


I have to think this has something to do with the Vendor ID. What I 
don't get is that opensips wouldn't load unless it was valid in the 
dictionary to begin with..


Any ideas?

Thanks,
Brett



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Re: [OpenSIPS-Users] db_mysql core dump

[Bogdan-Andrei Iancu]

Hi Duane,

yes, that is...but fortunately does not provide the information I was 
hoping for


Is it ok if I will send you a patch that will enable kind of hunting for 
this bug ?


Regards,
Bogdan

Duane Larson wrote:

I believe this is what you wanted
 
(gdb) frame 2
#2  0x7f176faf8f22 in db_mysql_delete (_h=0x812f20, 
_k=0x7fff61728980, _o=0x7fff61728960, _v=0x7fff61728900, _n=2) at 
dbase.c:893
893 ret = db_mysql_do_prepared_query(_h, 
&query_holder, _v, _n, NULL, 0);


(gdb) print _h
$2 = (const db_con_t *) 0x812f20

(gdb) print _h->curr_ps
$3 = (db_ps_t *) 0x7f176c46d2b0

(gdb) print (struct prep_stmt*)(*(_h)->curr_ps)
$4 = (struct prep_stmt *) 0x813650

(gdb) print ((struct prep_stmt*)(*(_h)->curr_ps))->stmts
$5 = (struct my_stmt_ctx *) 0x814d10

(gdb) print ((struct prep_stmt*)(*(_h)->curr_ps))->stmts->table.s
$6 = 0x814d48 "watchersdelete from watchers where inserted_timestatus=?.26705.61.7"


(gdb) print ((struct prep_stmt*)(*(_h)->curr_ps))->stmts->query.s
$7 = 0x814d50 "delete from watchers where inserted_timestatus=?.26705.61.7"


On Wed, Nov 24, 2010 at 5:29 AM, Bogdan-Andrei Iancu 
mailto:bog...@voice-system.ro>> wrote:


Hi Duane,

in frame 2 print the followings:
  _h
  _h->curr_ps
  (struct prep_stmt*)(*(_h)->curr_ps)
  ((struct prep_stmt*)(*(_h)->curr_ps))->stmts
  ((struct prep_stmt*)(*(_h)->curr_ps))->stmts->table.s
  ((struct prep_stmt*)(*(_h)->curr_ps))->stmts->query.s


Thanks and regards,
Bogdan



duane.lar...@gmail.com  wrote:

I searched the mailing list and couldn't find anything related
to this.

For the last couple of nights OpenSIPS has died on me. Each
time in Syslog I see

Nov 13 02:54:25 Proxy01 kernel: [1750051.944109]
opensips[6645]: segfault at 0 ip 7f7fd431bafd sp
7fff67048fc0 error 6 in db_mysql.so[7f7fd4311000+d
000]
Nov 13 02:54:25 Proxy01 ./opensips[6654]:
CRITICAL:core:receive_fd: EOF on 24
Nov 13 02:54:25 Proxy01 ./opensips[6631]:
INFO:core:handle_sigs: child process 6645 exited by a signal 11
Nov 13 02:54:25 Proxy01 ./opensips[6631]:
INFO:core:handle_sigs: core was generated
Nov 13 02:54:25 Proxy01 ./opensips[6631]:
INFO:core:handle_sigs: terminating due to SIGCHLD


The backtrace has the following

Core was generated by `./opensips -f
/usr/local/etc/opensips/opensips.cfg'.
Program terminated with signal 11, Segmentation fault.
#0 0x7f7fd431bafd in db_mysql_val2bind (v=0x7fff67049120,
binds=0x7ffd68, i=112) at val.c:274
274 *(binds[i].is_null) = 0;
(gdb) backtrace
#0 0x7f7fd431bafd in db_mysql_val2bind (v=0x7fff67049120,
binds=0x7ffd68, i=112) at val.c:274
#1 0x7f7fd431607a in db_mysql_do_prepared_query
(conn=0x7ff5f8, query=0x7f7fd452e6d0, v=0x7fff67049100, n=2,
uv=0x0, un=0) at dbase.c:443
#2 0x7f7fd4318305 in db_mysql_delete (_h=0x7ff5f8,
_k=0x7fff67049180, _o=0x7fff67049160, _v=0x7fff67049100, _n=2)
at dbase.c:893
#3 0x7f7fd0a96b10 in msg_watchers_clean (ticks=, param=) at subscribe.c:484
#4 0x0049e3da in timer_ticker () at timer.c:325
#5 run_timer_process () at timer.c:395
#6 start_timer_processes () at timer.c:475
#7 0x0042be57 in main_loop (argc=, argv=0x7fff67049378) at main.c:867
#8 main (argc=, argv=0x7fff67049378) at
main.c:1388


If need be I could also look in the MySQL logs and see what
the last execution was. Any ideas?




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Re: [OpenSIPS-Users] BYE processing and AVP

[Anton Zagorskiy]

Ok, thanks!

 

 

 

 


2



 

 

From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
Sent: Tuesday, November 30, 2010 5:29 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP

 

 

On Tue, Nov 30, 2010 at 11:19 PM, mayamatakeshi 
wrote:

 

2010/11/30 Anton Zagorskiy 

 

Why BYE isn't a part of the transaction?


>From RFC3261:

Specifically, a SIP transaction consists of a single request and any
responses to
   that request, which include zero or more provisional responses and
   one or more final responses.

But when opensips docs say transaction, it is also implying that you are
using module tm (if you don't use tm functions, then you are doing stateless
processing and no AVP will be saved).  


For completeness,
the BYE is part of the dialog established by the INVITE transaction that
ended when "200 OK" was received. But the BYE itself starts another
transaction.
So, the AVPs you set in the processing of the INVITE will be available (as
long as you use module tm) when your process responses for that INVITE (180,
183, 200 etc), but not when you are handling other requests in the dialog.
 

 


WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593

a.zagors...@oyster-telecom.ru

www.oyster-telecom.ru


From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
Sent: Tuesday, November 30, 2010 4:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP



On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
 wrote:
Hello.

During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:

if (has_totag())
{
   if (loose_route())
   {
   if (is_method("BYE"))
   {
   xlog("*** BYE $avp(i:2)");

>From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):

"AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present. "

br,
takeshi




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Re: [OpenSIPS-Users] BYE processing and AVP

[mayamatakeshi]

On Tue, Nov 30, 2010 at 11:19 PM, mayamatakeshi wrote:

>
> 2010/11/30 Anton Zagorskiy 
>
> Why BYE isn't a part of the transaction?
>>
>
> From RFC3261:
>
> Specifically, a SIP transaction consists of a single request and any 
> responses to
>that request, which include zero or more provisional responses and
>one or more final responses.
>
> But when opensips docs say transaction, it is also implying that you are
> using module tm (if you don't use tm functions, then you are doing stateless
> processing and no AVP will be saved).
>

For completeness,
the BYE is part of the dialog established by the INVITE transaction that
ended when "200 OK" was received. But the BYE itself starts another
transaction.
So, the AVPs you set in the processing of the INVITE will be available (as
long as you use module tm) when your process responses for that INVITE (180,
183, 200 etc), but not when you are handling other requests in the dialog.


>
>
>> WBR, Anton Zagorskiy
>> VoIP Developer, Oyster Telecom
>> Phone.: +7 812 601-0666
>> Fax: +7 812 601-0593
>> a.zagors...@oyster-telecom.ru
>> www.oyster-telecom.ru
>>
>>
>> From: users-boun...@lists.opensips.org
>> [mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
>> Sent: Tuesday, November 30, 2010 4:13 PM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] BYE processing and AVP
>>
>>
>> On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
>>  wrote:
>> Hello.
>>
>> During INVITE processing I'm setting up some avp variables. When I'm
>> receiving a BYE request all avps are null. Why does this happen?
>> A part of the script:
>>
>> if (has_totag())
>> {
>>if (loose_route())
>>{
>>if (is_method("BYE"))
>>{
>>xlog("*** BYE $avp(i:2)");
>>
>> From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):
>>
>> "AVPs are part of the transaction context, so they will be visible
>> everywhere where the transaction is present. "
>>
>> br,
>> takeshi
>>
>>
>>
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>>
>
>
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Re: [OpenSIPS-Users] BYE processing and AVP

[mayamatakeshi]

2010/11/30 Anton Zagorskiy 

> Why BYE isn't a part of the transaction?
>

>From RFC3261:

Specifically, a SIP transaction consists of a single request and any
responses to
   that request, which include zero or more provisional responses and
   one or more final responses.

But when opensips docs say transaction, it is also implying that you are
using module tm (if you don't use tm functions, then you are doing stateless
processing and no AVP will be saved).


> WBR, Anton Zagorskiy
> VoIP Developer, Oyster Telecom
> Phone.: +7 812 601-0666
> Fax: +7 812 601-0593
> a.zagors...@oyster-telecom.ru
> www.oyster-telecom.ru
>
>
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
> Sent: Tuesday, November 30, 2010 4:13 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] BYE processing and AVP
>
>
> On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
>  wrote:
> Hello.
>
> During INVITE processing I'm setting up some avp variables. When I'm
> receiving a BYE request all avps are null. Why does this happen?
> A part of the script:
>
> if (has_totag())
> {
>if (loose_route())
>{
>if (is_method("BYE"))
>{
>xlog("*** BYE $avp(i:2)");
>
> From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):
>
> "AVPs are part of the transaction context, so they will be visible
> everywhere where the transaction is present. "
>
> br,
> takeshi
>
>
>
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Re: [OpenSIPS-Users] BYE processing and AVP

[Anton Zagorskiy]

Why BYE isn't a part of the transaction?






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru


From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of mayamatakeshi
Sent: Tuesday, November 30, 2010 4:13 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] BYE processing and AVP


On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy
 wrote:
Hello.

During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:

if (has_totag())
{
       if (loose_route())
       {
               if (is_method("BYE"))
               {
                       xlog("*** BYE $avp(i:2)");

From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):

"AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present. "

br,
takeshi



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Re: [OpenSIPS-Users] BYE processing and AVP

[mayamatakeshi]

On Tue, Nov 30, 2010 at 10:07 PM, Anton Zagorskiy <
a.zagors...@oyster-telecom.ru> wrote:

> Hello.
>
> During INVITE processing I'm setting up some avp variables. When I'm
> receiving a BYE request all avps are null. Why does this happen?
> A part of the script:
>
> if (has_totag())
> {
>if (loose_route())
>{
>if (is_method("BYE"))
>{
>xlog("*** BYE $avp(i:2)");
>

>From the docs (http://www.opensips.org/Resources/DocsCoreVar#toc2):

"AVPs are part of the transaction context, so they will be visible
everywhere where the transaction is present. "

br,
takeshi
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[OpenSIPS-Users] BYE processing and AVP

[Anton Zagorskiy]

Hello.

During INVITE processing I'm setting up some avp variables. When I'm
receiving a BYE request all avps are null. Why does this happen?
A part of the script:

if (has_totag())
{
if (loose_route())
{
if (is_method("BYE"))
{
xlog("*** BYE $avp(i:2)");






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru





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Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

[Tijmen de Mes]

Hi,

On Nov 25, 2010, at 10:06 AM, ha do wrote:

and opensips.cfg with fifo config
# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0600)
modparam("mi_fifo", "fifo_group", 0)
modparam("mi_fifo", "fifo_group", "root")
modparam("mi_fifo", "fifo_user", 0)
modparam("mi_fifo", "fifo_user", "root")

and the callcontrol config.ini
[CallControl]
group   = root
[OpenSIPS]
socket_path = /tmp
;max_connections = 10 -- does it disconnect 10 calls in 1 time, if i  
need 100

terminated i need to increase to 100, right???

how to fix the issue
Try  using socket_path=/tmp/socket in config.ini. The socket path  
should match the "mi_datagram" ,"socket_name"

 in the opensips config.

Tijmen de Mes
AG Projects


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Re: [OpenSIPS-Users] OpenXCAP user authentication format

[Saúl Ibarra Corretgé]

On 11/28/2010 06:47 AM, duane.lar...@gmail.com wrote:

I am trying to get Bria and Snom to use the same XCAP documents. I am
having issues getting to play well with together with OpenXCAP.

Currently the only way to get Bria xcap config to authenticate with
OpenXCAP you have to manually set the username to be the OpenSIPS
subscriber. If you tell Bria to "Use SIP credentials" OpenXCAP thinks
the username is actually "sip" because it is taking the "sip" from the
uri "sip:...@xyz.com". Yet for some reason when a Snom phone sends
its username info OpenXCAP takes the whole sip uri as the username
successfully, but the authentication still fails because no user exists
that has the full uri "sip:x...@xyz.com"



OpenXCAP uses HTTP digest authentication mechanism (by default) so after 
it answers with the 401, specifying the realm and nonce, it's the client 
application the one that will take the username, password, realm, nonce, 
etc and answer with that hash. 
(http://en.wikipedia.org/wiki/Digest_access_authentication#Overview)


If you are using the integrated mode with OpenSIPS and not using 
cleartext passwords, HA1 values are stored, not passwords.



I am wondering what can be done when it comes to OpenXCAP user
authentication. Below you will see 3 displays.



Well, it's the user providing the wrong credentials, what do you expect 
OpenXCAP to do here?


I didn't have the time to test a Snom properly though I do have one and 
will test it when time allows :-)



Regards,

--
Saúl Ibarra Corretgé
AG Projects

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Re: [OpenSIPS-Users] Timer Based Failover Question

[Bruce Borrett]

Thank you Dave, this has been working perfectly for me.

Regards,
Bruce






From: Brett Nemeroff 
To: OpenSIPS users mailling list 
Sent: Tue, 23 November, 2010 23:31:09
Subject: Re: [OpenSIPS-Users] Timer Based Failover Question

I do this.. Works really well..
-Brett



On Tue, Nov 16, 2010 at 12:49 PM, Dave Singer  wrote:

I ran into the same problem with one of our carriers. The way I did it, with 
advice from bogdan, was to set the fr_inv_timer_avp to 6 sec (so long because 
of 
some calls to cell phone systems have long delays) when sending to the 
particular carrier then in a reply route special to that carrier, reset it to 
the normal 300 if the response was a 18x. Here are the pertinent parts:
>
>
>modparam("tm", "fr_inv_timer", 300) # Timer on Final response: 
>Minimum is 2 sec, Default is 120 sec.
>modparam("tm", "restart_fr_on_each_reply", 1)   # Reset fr_int_timer on each 
>reply. Needed if you want to adjust the fr_inv_timer_avp with avp depending on 
>reply.
>modparam("tm", "fr_inv_timer_avp", "$avp(i:2)") # Used if overide of 
>fr_inv_timer param is needed.
>modparam("tm", "onreply_avp_mode", 1)   # set to 1 if you want to 
>access 
>and or save avps from or for other parts of the transaction, like changing 
>the fr_inv_timer_avp.
>
>
>route[carrier_c] {
>if (is_method("INVITE")) {
>t_on_failure("2");
>t_on_reply("2");
>$avp(i:2) = 6;
>}
>}
>onreply_route[2] {
>/* once we get ring progress let it ring for upto 300 sec */
>fix_nated_contact();
>if ( $rs =~ "18." ) {
>$avp(i:2) = 300;
>#xlog("got ringing, reset final timer to $avp(i:2) sec.\n");
>}
>}
>
>
>On Tue, Nov 16, 2010 at 2:49 AM, Denis Putyato  wrote:
>
>And what about
>>http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id250384
>> 
>> 
>>From:users-boun...@lists.opensips.org 
>>[mailto:users-boun...@lists.opensips.org] 
>>On Behalf Of Bruce Borrett
>>Sent: Tuesday, November 16, 2010 1:40 PM
>>To: Users@lists.opensips.org
>>Subject: [OpenSIPS-Users] Timer Based Failover Question
>> 
>>Hi All
>>
>>I am having a problem where a SIP provider are sometimes sending us a 100, 
>>but 
>>then nothing afterwards. I would like to fail these calls over using a timer, 
>>but fr_timer wont work since we are receiving a 100, and fr_inv_timer 
>>requires a 
>>very lengthy duration which also will not work as I would like for the call 
>>to 
>>failover within 5 seconds maximum.
>>
>>Does anyone have any other suggestion for me please?
>>
>>Regards,
>>Bruce Borrett
>> 
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>>
>
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