[OpenSIPS-Users] is this valid naming for avps - $avp($var(i))

[Darren DeLitizia]

Is this valid naming for avp's - $avp($var(i))

I am trying to loop through exec_avp results as I will never know how many
rows are being returned. It is in a startup route and I am using a while
statement to loop through. It is not recognizing naming. $avp($avp(i))
always returns NULL while something like $avp(2) will return the data

Tx
DD

  exec_avp("/usr/bin/php /usr/local/etc/opensips/name_of_script.php
ips");#gets list of ips and other carrier info
  if (is_avp_set("$avp(1)")) {
$var(x) = $avp(1); #$avp(1) is the amount of rows returned so I know
how many to loop through
$var(x) = $(var(x){s.int}) + 1;
avp_delete("$avp(1)");
$avp(i) = 2;
while($avp(i) <= $var(x)) {
  xlog("TEST -- $avp(i)\n");
  xlog("TEST -- $avp($avp(i))\n"); #
  $avp(ip) = $(avp($var(i)){s.select,0,;});
  $avp(carrier_ids) = $(avp($var(i)){s.select,1,;});
  $avp(route_types) = $(avp($var(i)){s.select,2,;});
  $avp(carrier_names) = $(avp($var(i)){s.select,0,;});
  cache_store("local", "ips_carrier_ids_$avp(ip)",
"$avp(carrier_ids)");#
  cache_store("local", "ips_carrier_names_$avp(ip)",
"$avp(carrier_names)");#
  cache_store("local", "ips_carrier_types_$avp(ip)",
"$avp(route_types)");#
  avp_delete("$avp($var(i))");
  $avp(i) = $avp(i) + 1;
}
  }
}
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] grp, and dr_grp

[Duane Larson]

Here is an example from my test database

Proxy02:~# opensipsctl db show grp | grep 9012732009
3   9012732009  irock.com   int 2010-10-25 13:41:57
285 9012732009  irock.com   ld  2011-03-24 19:35:01
284 9012732009  irock.com   local   2011-03-24 19:34:49

Proxy02:~# opensipsctl db show grp | grep 9012732005
30  9012732005  coolbeans.com   int 2011-03-10 22:04:43
28  9012732005  coolbeans.com   ld  2011-03-10 22:04:27
20  9012732005  coolbeans.com   local   2010-11-05 15:52:08


Thats just right.  Your grp within the grp table is probably wrong.

Once again for droute I have no experience.



On Thu, Dec 15, 2011 at 8:31 PM, Nick Khamis  wrote:

> Thank you so much for your time Duane, and grp in that context makes
> perfect sense however,
> the username is really throwing me off. What I have now is:
>
>
> ++--+--+-+-+
> | id | username | domain   | grp | last_modified
> |
>
> ++--+--+-+-+
> | 38 | 1001 | opensips1.test.com | Local Group 1001 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
> | 39 | 1001 | opensips1.test.com | LD Group 1001 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
> | 40 | 1001 | opensips1.test.com | INT Group 1001 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
> | 41 | 1002 | opensips1.test.com | Local Group 1002 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
> | 42 | 1002 | opensips1.test.com | LD Group 1002 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
> | 43 | 1002 | opensips1.test.com | LD Group 1002 | 2011-12-15
> 20:21:24 |
>
> ++--+--+-+-+
>
> And that's just wrong..  As for droute, the grp table is not even used.
>
> Kind Regards,
>
> Nick.
>
>
>
>
>
> On Thu, Dec 15, 2011 at 9:12 PM, Duane Larson 
> wrote:
> > I can't help you when it comes to DR since I haven't had to work with it
> > yet.  As for the grp you can think of it like the normal examples I have
> > seen (group local is for users that will be allowed to make local calls,
> > group ld is for users that can make long distance calls and group int are
> > for users that can make international calls).  That's one way to use grp.
> >
> > On Dec 15, 2011 7:46 PM, "Nick Khamis"  wrote:
> >>
> >> I am just trying to understand how to assign a specific group to
> >> specific dialplans, and having a hard time
> >> getting it right. Maybe an example?
> >>
> >> grp Table
> >>
> >>
> ++--+--+-+-+
> >> | id | username | domain   | grp | last_modified
> >> |
> >>
> >>
> ++--+--+-+-+
> >> | 38 | 1001 | opensips1.test.com | Local Group | 2011-12-15
> 20:21:24 |
> >>
> >>
> ++--+--+-+-+
> >> | 39 | 1002 | opensips1.test.com | Local Group | 2011-12-15
> 20:21:24 |
> >>
> >>
> ++--+--+-+-+
> >> | 40 | 1003 | opensips1.test.com | Local Group | 2011-12-15
> 20:21:24 |
> >>
> >>
> ++--+--+-+-+
> >>
> >>
> >> The username, and domain field is my first confusion. For some reason
> >> I would think the this table
> >> would have a subscriber_id that is a foreign key to the subsriber
> >> table? This would be "group table",
> >> and name would be something like "Local Group", "VoiceMail Group"
> >>
> >> dr_groups Table
> >>
> >> ++--+--+-++
> >> | id | username | domain   | groupid | description|
> >> ++--+--+-++
> >> | 42 | 1001 | opensips1.test.com |   0 | Access Granted |
> >> ++--+--+-++
> >> | 43 | 1002 | opensips1.test.com |   0 | Access Granted |
> >> ++--+--+-++
> >> | 43 | 1003 | opensips1.test.com |   0 | Access Granted |
> >> ++--+--+-++
> >>
> >>
> >> Same table?
> >>
> >> dr_rules;
> >>
> >>
> ++-++-+--+-++---++
> >> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
> >> attrs | description|
> >>
> >>
> ++-++-+--+---

Re: [OpenSIPS-Users] grp, and dr_grp

[Nick Khamis]

Thank you so much for your time Duane, and grp in that context makes
perfect sense however,
the username is really throwing me off. What I have now is:

++--+--+-+-+
| id | username | domain   | grp | last_modified   |
++--+--+-+-+
| 38 | 1001 | opensips1.test.com | Local Group 1001 | 2011-12-15 20:21:24 |
++--+--+-+-+
| 39 | 1001 | opensips1.test.com | LD Group 1001 | 2011-12-15 20:21:24 |
++--+--+-+-+
| 40 | 1001 | opensips1.test.com | INT Group 1001 | 2011-12-15 20:21:24 |
++--+--+-+-+
| 41 | 1002 | opensips1.test.com | Local Group 1002 | 2011-12-15 20:21:24 |
++--+--+-+-+
| 42 | 1002 | opensips1.test.com | LD Group 1002 | 2011-12-15 20:21:24 |
++--+--+-+-+
| 43 | 1002 | opensips1.test.com | LD Group 1002 | 2011-12-15 20:21:24 |
++--+--+-+-+

And that's just wrong..  As for droute, the grp table is not even used.

Kind Regards,

Nick.





On Thu, Dec 15, 2011 at 9:12 PM, Duane Larson  wrote:
> I can't help you when it comes to DR since I haven't had to work with it
> yet.  As for the grp you can think of it like the normal examples I have
> seen (group local is for users that will be allowed to make local calls,
> group ld is for users that can make long distance calls and group int are
> for users that can make international calls).  That's one way to use grp.
>
> On Dec 15, 2011 7:46 PM, "Nick Khamis"  wrote:
>>
>> I am just trying to understand how to assign a specific group to
>> specific dialplans, and having a hard time
>> getting it right. Maybe an example?
>>
>> grp Table
>>
>> ++--+--+-+-+
>> | id | username | domain               | grp         | last_modified
>> |
>>
>> ++--+--+-+-+
>> | 38 | 1001     | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>>
>> ++--+--+-+-+
>> | 39 | 1002     | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>>
>> ++--+--+-+-+
>> | 40 | 1003     | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>>
>> ++--+--+-+-+
>>
>>
>> The username, and domain field is my first confusion. For some reason
>> I would think the this table
>> would have a subscriber_id that is a foreign key to the subsriber
>> table? This would be "group table",
>> and name would be something like "Local Group", "VoiceMail Group"
>>
>> dr_groups Table
>>
>> ++--+--+-++
>> | id | username | domain               | groupid | description    |
>> ++--+--+-++
>> | 42 | 1001     | opensips1.test.com |       0 | Access Granted |
>> ++--+--+-++
>> | 43 | 1002     | opensips1.test.com |       0 | Access Granted |
>> ++--+--+-++
>> | 43 | 1003     | opensips1.test.com |       0 | Access Granted |
>> ++--+--+-++
>>
>>
>> Same table?
>>
>> dr_rules;
>>
>> ++-++-+--+-++---++
>> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
>> attrs | description        |
>>
>> ++-++-+--+-++---++
>> |     16 | 0       |        |         |        1 | 0       | 9      |
>> yes   | ASTCluster Gateway |
>>
>> ++-++-+--+-++---++
>>
>> Is the groupid field of this table a FK to the grp or dr_groups table?
>>
>> dialplan;
>>
>> ++--++--+--+---+---+--+---+
>> | id | dpid | pr | match_op | match_exp                        |
>> match_len | subst_exp | repl_exp | attrs |
>>
>> ++--++--+--+---+---+--+---+
>> | 24 |    0 |  1 |        1 | ^[2-9][0-9]{6}.*                 |
>>  0 |           |          | local |
>> | 28 |    0 |  1 |        0 | *98                              |
>>  0 |           |          | voicmail |
>> | 29 |    1 |  1 |        0 | *98

Re: [OpenSIPS-Users] grp, and dr_grp

[Duane Larson]

I can't help you when it comes to DR since I haven't had to work with it
yet.  As for the grp you can think of it like the normal examples I have
seen (group local is for users that will be allowed to make local calls,
group ld is for users that can make long distance calls and group int are
for users that can make international calls).  That's one way to use grp.
On Dec 15, 2011 7:46 PM, "Nick Khamis"  wrote:

> I am just trying to understand how to assign a specific group to
> specific dialplans, and having a hard time
> getting it right. Maybe an example?
>
> grp Table
>
> ++--+--+-+-+
> | id | username | domain   | grp | last_modified
> |
>
> ++--+--+-+-+
> | 38 | 1001 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>
> ++--+--+-+-+
> | 39 | 1002 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>
> ++--+--+-+-+
> | 40 | 1003 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
>
> ++--+--+-+-+
>
>
> The username, and domain field is my first confusion. For some reason
> I would think the this table
> would have a subscriber_id that is a foreign key to the subsriber
> table? This would be "group table",
> and name would be something like "Local Group", "VoiceMail Group"
>
> dr_groups Table
>
> ++--+--+-++
> | id | username | domain   | groupid | description|
> ++--+--+-++
> | 42 | 1001 | opensips1.test.com |   0 | Access Granted |
> ++--+--+-++
> | 43 | 1002 | opensips1.test.com |   0 | Access Granted |
> ++--+--+-++
> | 43 | 1003 | opensips1.test.com |   0 | Access Granted |
> ++--+--+-++
>
>
> Same table?
>
> dr_rules;
>
> ++-++-+--+-++---++
> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
> attrs | description|
>
> ++-++-+--+-++---++
> | 16 | 0   || |1 | 0   | 9  |
> yes   | ASTCluster Gateway |
>
> ++-++-+--+-++---++
>
> Is the groupid field of this table a FK to the grp or dr_groups table?
>
> dialplan;
>
> ++--++--+--+---+---+--+---+
> | id | dpid | pr | match_op | match_exp|
> match_len | subst_exp | repl_exp | attrs |
>
> ++--++--+--+---+---+--+---+
> | 24 |0 |  1 |1 | ^[2-9][0-9]{6}.* |
>  0 |   |  | local |
> | 28 |0 |  1 |0 | *98  |
>  0 |   |  | voicmail |
> | 29 |1 |  1 |0 | *98  |
>  0 |   |  | voicmail |
>
> ++--++--+--+---+---+--+---+
>
> What I am trying to accomplish is:
>
> Local Group (dpid 0)
>+
>| _ _ _ 1001 test.com password (From Subscriber)
>
>
> VoiceMail Group (dpid 1)
>+
>| _ _ _ 1002 test.com password (From Subscriber)
>   +
>| _ _ _ 1003 test.com password (From Subscriber)
>
>
> Also, I would like to use this module to perform LCR (country code,
> area code, rate etc..).
> Is this possible instead of using the LCR module?
>
> Thanks in Advance,
>
> Nick.
>
> ___
> 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] grp, and dr_grp

[Nick Khamis]

I am just trying to understand how to assign a specific group to
specific dialplans, and having a hard time
getting it right. Maybe an example?

grp Table
++--+--+-+-+
| id | username | domain   | grp | last_modified   |
++--+--+-+-+
| 38 | 1001 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
++--+--+-+-+
| 39 | 1002 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
++--+--+-+-+
| 40 | 1003 | opensips1.test.com | Local Group | 2011-12-15 20:21:24 |
++--+--+-+-+


The username, and domain field is my first confusion. For some reason
I would think the this table
would have a subscriber_id that is a foreign key to the subsriber
table? This would be "group table",
and name would be something like "Local Group", "VoiceMail Group"

dr_groups Table

++--+--+-++
| id | username | domain   | groupid | description|
++--+--+-++
| 42 | 1001 | opensips1.test.com |   0 | Access Granted |
++--+--+-++
| 43 | 1002 | opensips1.test.com |   0 | Access Granted |
++--+--+-++
| 43 | 1003 | opensips1.test.com |   0 | Access Granted |
++--+--+-++


Same table?

dr_rules;
++-++-+--+-++---++
| ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
attrs | description|
++-++-+--+-++---++
| 16 | 0   || |1 | 0   | 9  |
yes   | ASTCluster Gateway |
++-++-+--+-++---++

Is the groupid field of this table a FK to the grp or dr_groups table?

dialplan;
++--++--+--+---+---+--+---+
| id | dpid | pr | match_op | match_exp|
match_len | subst_exp | repl_exp | attrs |
++--++--+--+---+---+--+---+
| 24 |0 |  1 |1 | ^[2-9][0-9]{6}.* |
  0 |   |  | local |
| 28 |0 |  1 |0 | *98  |
  0 |   |  | voicmail |
| 29 |1 |  1 |0 | *98  |
  0 |   |  | voicmail |
++--++--+--+---+---+--+---+

What I am trying to accomplish is:

Local Group (dpid 0)
+
| _ _ _ 1001 test.com password (From Subscriber)


VoiceMail Group (dpid 1)
+
| _ _ _ 1002 test.com password (From Subscriber)
   +
| _ _ _ 1003 test.com password (From Subscriber)


Also, I would like to use this module to perform LCR (country code,
area code, rate etc..).
Is this possible instead of using the LCR module?

Thanks in Advance,

Nick.

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

Re: [OpenSIPS-Users] grp, and dr_grp

[Duane Larson]

That is the only way you could add a user to a group.
On Dec 15, 2011 3:39 PM, "Nick Khamis"  wrote:

> Why the username field?
>
> Cheers,
>
> Nick.
>
> ___
> 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] LCR using DROUTING Module

[Nick Khamis]

Hello Everyone,

We would like to use the DROUTING module to:
* Limit users to specific dialplans
* Perform LCR

Is there any step by step tutorial on how to do this?

Thanks in Advance,

Nick.

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

Re: [OpenSIPS-Users] uac_replace_from corruption

[Razvan Crainea]

Hi, Brett!

I was talking about the DEBUG info (debug core parameter set to 6). If 
you can provide the logs for such a corruption on pastebin.com or 
something it would be great. Otherwise I will try to reproduce myself 
this scenario and see what's happening.
I'm not sure what version are you using, but the dialog based version 
was only added in 1.7. In 1.6 this function was indeed using the vst/vsf 
headers, and if you don't see them any more in your traces it might mean 
that you somehow copied the uac module from 1.7 to 1.6. Please double 
check that.


Regards,
Răzvan

On 12/15/2011 11:40 PM, Brett Nemeroff wrote:

Razvan,
I'm using the new dialog based method. If I remember right the old way 
was with vsf/vst headers and I'm definitely not using that anymore.


The version I'm using is a svn pull from 1.6.3 (7919), which I think 
is how I'm doing the dialog based version. I'm not quite ready to move 
to 1.7 but I'm close.


These servers I'm having this issue on have quite a bit of load on 
them; but I can try to isolate the issue. Here's some entries that are 
in my log that might help you out:


Dec 15 21:28:07 lcr02 /usr/local/sbin/opensips[1433]: 
ERROR:core:parse_from_header: bad msg or missing FROM header
Dec 15 21:28:07 lcr02 /usr/local/sbin/opensips[1433]: 
ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
Dec 15 21:30:55 lcr02 /usr/local/sbin/opensips[1432]: 
ERROR:core:parse_to: unexpected char [>] in status 10: 
<<> wrote:


Hi, Brett!

Are you using uac_replace_from function with dialog support or the
old approach with route parameters? Would it be possible to
increase OpenSIPS debugging level and paste some extra information?

Regards,

--
Răzvan Crainea
OpenSIPS Developer


On 12/15/2011 12:41 AM, Brett Nemeroff wrote:

Hello List,
I'm sure I brought this up sometime before, but I'm still having
this issue. I have to imagine that some of you out there are
having it as well. Running OpenSIP-1.6.4

I perform a uac_replace_from. It works just fine, But on
subsequent transactions within the dialog, like a BYE that comes
from the destination, the restored From is corrupted IF there was
a port in the original From. This appears to be isolated to cases
where the URI has a port in it.

For example, here's the From from the original INVITE
From: "8005551212 " http://sip:8005551212@1.2.3.4:5070>>;tag=as3371c611.

And then the far end sends a BYE later on:
From: "18705551313 " mailto:sip%3A18705551313@2.3.4.5>>;tag=SDsjh0f99-eb402829

And I relay the BYE:
From: "18705551313 " mailto:sip:18705551313@14.5><4)2$6:^E7>;tag=SDsjh0f99-eb402829

So aside from getting the client to stop sending the port in the
From header, is there another fix for this?

Thanks!
-Brett



___
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


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

Re: [OpenSIPS-Users] Opensips with Heartbeat

[Schneur Rosenberg]

thanks your advice solved the problem

On Fri, Dec 16, 2011 at 1:12 AM, Ryan Bullock  wrote:
> Opensips is trying to do a reverse dns on the IP address. You can try
> adding a reverse dns record for that ip or try disabling the dns
> and/or auto_aliases options.
>
>
> On Thu, Dec 15, 2011 at 2:52 PM, Schneur Rosenberg
>  wrote:
>> I'm using opensips on a computer with 2 ip addresses one steady one
>> and one is a floating ip address provided by heartbeat, when heartbeat
>> is on and I have 2 ip addresses opensips takes a very long time to
>> start and I get a error in the messages file, the error is
>>
>>  opensips: WARNING:core:fix_socket_list: could not rev. resolve 64.69.47.245
>>
>> where 64.69.47.245 is the floating ip
>>
>> when I remove the floating ip address everything works smooth.
>>
>> thanks in advance
>> S. Rosenberg
>>
>> ___
>> 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

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

Re: [OpenSIPS-Users] Opensips with Heartbeat

[Ryan Bullock]

Opensips is trying to do a reverse dns on the IP address. You can try
adding a reverse dns record for that ip or try disabling the dns
and/or auto_aliases options.


On Thu, Dec 15, 2011 at 2:52 PM, Schneur Rosenberg
 wrote:
> I'm using opensips on a computer with 2 ip addresses one steady one
> and one is a floating ip address provided by heartbeat, when heartbeat
> is on and I have 2 ip addresses opensips takes a very long time to
> start and I get a error in the messages file, the error is
>
>  opensips: WARNING:core:fix_socket_list: could not rev. resolve 64.69.47.245
>
> where 64.69.47.245 is the floating ip
>
> when I remove the floating ip address everything works smooth.
>
> thanks in advance
> S. Rosenberg
>
> ___
> 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] Opensips with Heartbeat

[Schneur Rosenberg]

I'm using opensips on a computer with 2 ip addresses one steady one
and one is a floating ip address provided by heartbeat, when heartbeat
is on and I have 2 ip addresses opensips takes a very long time to
start and I get a error in the messages file, the error is

 opensips: WARNING:core:fix_socket_list: could not rev. resolve 64.69.47.245

where 64.69.47.245 is the floating ip

when I remove the floating ip address everything works smooth.

thanks in advance
S. Rosenberg

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

Re: [OpenSIPS-Users] all sip body headers regarding video removed

[Cindy Leung]

Hello,

I have to delete codec for similar reasons.  I'm trying to use 
codec_delete_except_re().

Here's the incoming SDP:
m=audio 49152 RTP/AVP 123 122 125 121 124 9 0 8 113 100
a=sendrecv
a=ptime:20
a=rtpmap:123 G7221/32000
a=fmtp:123 bitrate=48000
a=rtpmap:122 G7221/32000
a=fmtp:122 bitrate=32000
a=rtpmap:125 G7221/16000
a=fmtp:125 bitrate=32000
a=rtpmap:121 G7221/32000
a=fmtp:121 bitrate=24000
a=rtpmap:124 G7221/16000
a=fmtp:124 bitrate=24000
a=rtpmap:113 MP4A-LATM/32000
a=rtpmap:100 telephone-event/8000
m=video 49154 RTP/AVP 98 34 96 97
a=sendrecv
a=rtpmap:98 H264/9
a=fmtp:98 profile-level-id=42801f; max-fs=21600
a=rtpmap:34 H263/9
a=fmtp:34 CIF=3;QCIF=2
a=rtpmap:96 H263-1998/9
a=fmtp:96 CIF4=1;CIF=1;QCIF=1;F=1;I=1;J=1;T=1
a=rtpmap:97 H263-2000/9
a=fmtp:97 profile=1; level=42
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 49158 RTP/AVP 110
a=sendrecv
a=rtpmap:110 H224/4800

... which then goes through 
codec_delete_except_re("PCMU|G729|telephone-event");

... and I end up getting
m=audio 28964 RTP/AVP 9 0 8 100
a=sendrecv
a=ptime:20
a=rtpmap:100 telephone-event/8000
m=video 28734 RTP/AVP 
a=sendrecv
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
m=application 27676 RTP/AVP 
a=sendrecv

Two issues I'm seeing here:
1.  Static audio payloads are not being removed (in this case G722 and G711a).  
However, if the a= lines are provided, then they get removed as expected.
2.  The m= and a= lines for "video" and "application" are still there while the 
payload types are all deleted.  

Thanks.


Cinthia



On Aug 4, 2011, at 11:41 AM, Dani Popa wrote:

> Thanks,
> 
> I already did that.
> 
> Dani
> 
> On 08/04/11 18:19, Razvan Crainea wrote:
>> Hi Dani,
>> 
>> You can try by deleting the most common video codecs (like H261, H263, H264).
>> You can do that using the codec_delete[1] functions from the textops module.
>> I think you should also replace the video port with 0.
>> 
>> [1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910
>> 
>> Regards,
>> 
>> Razvan Crainea
>> OpenSIPS Developer
>> 
>> 
>> On 04.08.2011 18:03, Dani Popa wrote:
>>> Hi,
>>> 
>>> In fact, i have some problems with one of my pstn gw's that send "400 
>>> Incorrect content length", i think, because of too long sip packet. So, 
>>> because it is pstn, i want to remove video capability(many lines in first 
>>> invite packet).
>>> 
>>> Dani
>>> 
>>> On 08/04/11 17:02, Razvan Crainea wrote:
 Hi Dani,
 
 Why would you do that? If you don't want to allow video, you can simply 
 replace the video port in the "m=" line with 0.
 
 Regards,
 
 Razvan Crainea
 OpenSIPS Developer
 
 
 On 04.08.2011 16:58, Dani Popa wrote:
> Hi all,
> 
> How can i remove all sip video body headers regardin video. Should i 
> remove any line from body after "m=video", or how. Please give me a hint, 
> if you have.
> 
> Thanks,
> Dani
> 
> ___
> 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
 
>>> 
>>> ___
>>> 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
>> 
> 
> ___
> 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] uac_replace_from corruption

[Brett Nemeroff]

Razvan,
I'm using the new dialog based method. If I remember right the old way was
with vsf/vst headers and I'm definitely not using that anymore.

The version I'm using is a svn pull from 1.6.3 (7919), which I think is how
I'm doing the dialog based version. I'm not quite ready to move to 1.7 but
I'm close.

These servers I'm having this issue on have quite a bit of load on them;
but I can try to isolate the issue. Here's some entries that are in my log
that might help you out:

Dec 15 21:28:07 lcr02 /usr/local/sbin/opensips[1433]:
ERROR:core:parse_from_header: bad msg or missing FROM header
Dec 15 21:28:07 lcr02 /usr/local/sbin/opensips[1433]:
ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
Dec 15 21:30:55 lcr02 /usr/local/sbin/opensips[1432]: ERROR:core:parse_to:
unexpected char [>] in status 10: <<>> .
Dec 15 21:30:55 lcr02 /usr/local/sbin/opensips[1432]:
ERROR:core:parse_from_header: bad from header
Dec 15 21:30:55 lcr02 /usr/local/sbin/opensips[1432]:
ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr

What kind of debug would you need? I may be able to replace it on a smaller
scale.

Thanks!
-Brett

On Thu, Dec 15, 2011 at 4:25 AM, Razvan Crainea
wrote:

>  Hi, Brett!
>
> Are you using uac_replace_from function with dialog support or the old
> approach with route parameters? Would it be possible to increase OpenSIPS
> debugging level and paste some extra information?
>
> Regards,
>
> --
> Răzvan Crainea
> OpenSIPS Developer
>
>
> On 12/15/2011 12:41 AM, Brett Nemeroff wrote:
>
> Hello List,
> I'm sure I brought this up sometime before, but I'm still having this
> issue. I have to imagine that some of you out there are having it as well.
> Running OpenSIP-1.6.4
>
> I perform a uac_replace_from. It works just fine, But on subsequent
> transactions within the dialog, like a BYE that comes from the destination,
> the restored From is corrupted IF there was a port in the original From.
> This appears to be isolated to cases where the URI has a port in it.
>
> For example, here's the From from the original INVITE
> From: "8005551212" ;tag=as3371c611.
>
> And then the far end sends a BYE later on:
> From: "18705551313" ;tag=SDsjh0f99-eb402829
>
> And I relay the BYE:
> From: "18705551313"  <4)2$6:^E7>;tag=SDsjh0f99-eb402829
>
> So aside from getting the client to stop sending the port in the From
> header, is there another fix for this?
>
> Thanks!
> -Brett
>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ___
> 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] grp, and dr_grp

[Nick Khamis]

Why the username field?

Cheers,

Nick.

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

Re: [OpenSIPS-Users] uac_replace_from corruption

[Razvan Crainea]

Hello,

Jeff, I was wrong, this feature is not available in 1.6, only in 1.7 .
Anyway, according to Brett's traces it seems to be a problem with the 
function and I would really appreciate if he could help me debugging this.


Regards,
Ra(zvan

On 12/15/2011 07:27 PM, Jeff Pyle wrote:

Brett,

Is the other end an Acme?  If so, they need to implement some custom 
parameters (which I do not have) to//honor some parts of section 12 of 
RFC3261 in such a way that won't break uac_replace_from().  Let me 
know if this is the case and we'll talk more.



Rasvan,

Can you share more about the "new" way to do it with the dialog 
module?  Is this available in 1.6?




- Jeff


___
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] Users Digest, Vol 41, Issue 52

[magnusadil...@gmail.com]

 like the b2bua module is mangling the cancel message and
is
ripping out the callid when sending upstream:


U 2011/12/07 20:15:05.895915 192.168.1.143:5060 -> 192.168.1.145:5090

CANCEL sip:9993518045551212@192.168.1.145:5090 SIP/2.0.

Via: SIP/2.0/UDP 192.168.1.143;branch=z9hG4bKac0e.5a3d2bf1.0.

From: "8669800222" ;tag=3532277698-944952.

Call-ID: 494823-3532277698-944947@192.168.1.1.

To: "18045551212" .

CSeq: 1 CANCEL.

Max-Forwards: 70.

User-Agent: Opensips.

Content-Length: 0.

.



U 2011/12/07 20:15:05.896027 192.168.1.145:5090 -> 192.168.1.143:5060

SIP/2.0 200 canceling.

Via: SIP/2.0/UDP 192.168.1.143;branch=z9hG4bKac0e.5a3d2bf1.0.

From: "8669800222" ;tag=3532277698-944952.

Call-ID: 494823-3532277698-944947@192.168.1.1.

To: "18045551212"


;tag=3330ae74b9cf9aed85afbc9203dd6238-e6b7.

CSeq: 1 CANCEL.

Server: Opensips.

Content-Length: 0.

.



U 2011/12/07 20:15:05.896097 192.168.1.145:5090 -> 10.2.3.210:5060

CANCEL sip:+18045551212@65.211.120.23 SIP/2.0.

Via: SIP/2.0/UDP 192.168.1.145:5090;branch=z9hG4bK0299.252f8e61.0.

.

From: ;tag=418802140f6308e008db76a1e1de765b.

CSeq: 2 INVITE54.7172739.

Content-Lengt

To: sip:+18045551212@65.211.120.237.

Call- CANCEL.

User-Agent: OpenSIPS (1.7.1-notls (x86_64/linux)).

Max-Forwards: 70.

Init-CallID: 494823-3532277698-944947@192.168.1.1.

Contact: .

.



U 2011/12/07 20:15:05.910842 10.2.3.210:5060 -> 192.168.1.145:5090

SIP/2.0 400 Missing Mandatory Header Call-Id.

v: SIP/2.0/UDP
192.168.1.145:5090;branch=z9hG4bK0299.252f8e61.0;received=192.168.1.145.

l: 0.
  

  

  

  
  
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-- next part --
An HTML attachment was scrubbed...
URL: 

--

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


End of Users Digest, Vol 41, Issue 52
*


__ Información de ESET Smart Security, versión de la base de firmas de virus 6715 (20111215) __

ESET Smart Security ha comprobado este mensaje.

http://www.eset.com


  




__ Información de ESET Smart Security, versión de la base de firmas de virus 6715 (20111215) __

ESET Smart Security ha comprobado este mensaje.

http://www.eset.com





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

Re: [OpenSIPS-Users] B2BUA Ripping/Truncating Callid

[Ovidiu Sas]

It was not fixed.  It is the same bug.

Regards,
Ovidiu Sas

On Thu, Dec 15, 2011 at 3:20 PM, Logan  wrote:
> Out of curiosity, based on the feedback in this bug; is this something
> that's being fixed? I notice this bug was for 1.6.4, but my experience is in
> 1.7.1. so I want to make sure if this was fixed, I report a new bug for
> 1.7.1
>
>
> Hi Bogdan,
>
> This bug fix requires further work in tm module, in local_route processing,
> so as to update the shortcuts in tm when lumps are applied for headers
> also. The fix that was committed last week solved this problem only when
> body lumps were applied.
> Unfortunately, I don't have time to work on this, so I have removed the
> assignation to me for this bug report.
>
> Regards,
> Anca
>
>
> On Dec 12, 2011, at 09:05 AM, Ovidiu Sas  wrote:
>
> Yes, indeed. Thanks for pointing out.
>
> Regards,
> Ovidiu Sas
>
> On Mon, Dec 12, 2011 at 1:26 AM, Ryan Bullock  wrote:
>> I think this is related to a bug that is already open:
>>
>> http://sourceforge.net/tracker/?func=detail&aid=3316230&group_id=232389&atid=1086410
>>
>>
>> On Fri, Dec 9, 2011 at 5:46 PM, Ovidiu Sas  wrote:
>>> Yeah, it's the first request after the modified INVITE that is
>>> malformed (I reproduced this running a snapshot from trunk).  Please
>>> open a bug report.
>>>
>>> Regards,
>>> Ovidiu Sas
>>>
>>> --
>>> VoIP Embedded, Inc.
>>> http://www.voipembedded.com
>>>
>>>
>>> On Fri, Dec 9, 2011 at 1:52 PM, Logan  wrote:
 I added the log and everything looks fine. It's only adding the PAI to
 the
 initial invite which is what I want. The odd thing is there are no
 issues
 with the invites, it just looks like the Cancel messages that are being
 mangled. I posted a separate issue to the list prior to this report but
 no
 one responded, I'm not sure it went through correctly but resulting
 cancel
 coming out of the B2BUA looked like this:

  Reference:

 192.168.1.146 = Opensips Proxy
 192.168.1.145 = Opensips B2BUA
 10.2.3.245 = Carrier



 U 2011/12/01 22:51:11.558887 192.168.1.146:5060 -> 192.168.1.145:5090
 CANCEL sip:9993512125551212@192.168.1.145:5090 SIP/2.0.
 Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0.
 From: "James Logan" ;tag=as06eabdcd.
 Call-ID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137.
 To: "12125551212" .
 CSeq: 102 CANCEL.
 Max-Forwards: 70.
 User-Agent: Opensips.
 Content-Length: 0.
 .


 U 2011/12/01 22:51:11.559378 192.168.1.145:5090 -> 192.168.1.146:5060
 SIP/2.0 200 canceling.
 Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0.
 From: "James Logan" ;tag=as06eabdcd.
 Call-ID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137.
 To: "12125551212"

 ;tag=3330ae74b9cf9aed85afbc9203dd6238-715f
 CSeq: 102 CANCEL.
 Server: B2BUA.
 Content-Length: 0.
 .


 U 2011/12/01 22:51:11.559527 192.168.1.145:5090 -> 10.2.3.245:5060
 CANCEL i...i.. SIP/2.0.
 Via: SIP/2.0/UDP 192.168.1.145:5090;branch=z9hG4bK5421.22999dd2.0.

 B2B.256.3572553sip:+12125551212@10.2.3.245sip:888444@192.168.1.1379120d3`.p..i...q.i
  CANCEL.
 User-Agent: OpenSIPS (1.7.1-notls (x86_64/linux)).
 Max-Forwards: 70.
 User-Agent: Opensips.
 Init-CallID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137.
 Contact: .
 .

 On Dec 07, 2011, at 05:18 PM, Ovidiu Sas  wrote:

 Add a log and print out what are you adding before adding it and you
 will see if it's good or not.

 On Wed, Dec 7, 2011 at 5:13 PM, Logan  wrote:
> This is the extent of my local route. If the $var is not present, I do
> not
> add it. Do you see any issue with what I'm doing here?
>
>
> local_route {
>         #xlog("L_INFO","* IN LOCAL ROUTE \n");
>
>         if (is_method("INVITE")) {
>                 if($var(pai_userpart)) {
>                         append_hf("P-Asserted-Identity:
> \"$var(pai_display)\" \r\n");
>                 }else{
>                         xlog("L_INFO","PAI is not present, not
> adding\n");
>                 }
>         }
>
>
> }
>
> On Dec 07, 2011, at 04:57 PM, Ovidiu Sas  wrote:
>
> You need to be careful when you alter requests in B2B mode (the
> received INVITE and the sent INVITE belong to different transactions).
> Make sure that you have something valid in those vars before applying
> any changes to the outgoing message.
>
> Regards,
> Ovidiu Sas
>
> On Wed, Dec 7, 2011 at 4:49 PM, Logan  wrote:
>> I'm storing some $vars in route[0] prior to calling
>> b2b_init_request("top
>> hiding");
>>
>> Then in my local route Im appending a P-Asserted-Identity header.
>>
>> I can't us

Re: [OpenSIPS-Users] B2BUA Ripping/Truncating Callid

[Logan]

Out of curiosity, based on the feedback in this bug; is this something that's being fixed? I notice this bug was for 1.6.4, but my experience is in 1.7.1. so I want to make sure if this was fixed, I report a new bug for 1.7.1Hi Bogdan,

This bug fix requires further work in tm module, in local_route processing,
so as to update the shortcuts in tm when lumps are applied for headers
also. The fix that was committed last week solved this problem only when
body lumps were applied. 
Unfortunately, I don't have time to work on this, so I have removed the
assignation to me for this bug report.

Regards,
AncaOn Dec 12, 2011, at 09:05 AM, Ovidiu Sas  wrote:Yes, indeed. Thanks for pointing out.  Regards, Ovidiu Sas  On Mon, Dec 12, 2011 at 1:26 AM, Ryan Bullock  wrote: > I think this is related to a bug that is already open: > http://sourceforge.net/tracker/?func=detail&aid=3316230&group_id=232389&atid=1086410 > > > On Fri, Dec 9, 2011 at 5:46 PM, Ovidiu Sas  wrote: >> Yeah, it's the first request after the modified INVITE that is >> malformed (I reproduced this running a snapshot from trunk).  Please >> open a bug report. >> >> Regards, >> Ovidiu Sas >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> >> On Fri, Dec 9, 2011 at 1:52 PM, Logan  wrote: >>> I added the log and everything looks fine. It's only adding the PAI to the >>> initial invite which is what I want. The odd thing is there are no issues >>> with the invites, it just looks like the Cancel messages that are being >>> mangled. I posted a separate issue to the list prior to this report but no >>> one responded, I'm not sure it went through correctly but resulting cancel >>> coming out of the B2BUA looked like this: >>> >>>  Reference: >>> >>> 192.168.1.146 = Opensips Proxy >>> 192.168.1.145 = Opensips B2BUA >>> 10.2.3.245 = Carrier >>> >>> >>> >>> U 2011/12/01 22:51:11.558887 192.168.1.146:5060 -> 192.168.1.145:5090 >>> CANCEL sip:9993512125551212@192.168.1.145:5090 SIP/2.0. >>> Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0. >>> From: "James Logan" 888444@192.168.1.137>;tag=as06eabdcd. >>> Call-ID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137. >>> To: "12125551212" 12125551212@192.168.1.146>. >>> CSeq: 102 CANCEL. >>> Max-Forwards: 70. >>> User-Agent: Opensips. >>> Content-Length: 0. >>> . >>> >>> >>> U 2011/12/01 22:51:11.559378 192.168.1.145:5090 -> 192.168.1.146:5060 >>> SIP/2.0 200 canceling. >>> Via: SIP/2.0/UDP 192.168.1.146;branch=z9hG4bK2df7.78db1d81.0. >>> From: "James Logan" 888444@192.168.1.137>;tag=as06eabdcd. >>> Call-ID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137. >>> To: "12125551212" >>> 12125551212@192.168.1.146>;tag=3330ae74b9cf9aed85afbc9203dd6238-715f >>> CSeq: 102 CANCEL. >>> Server: B2BUA. >>> Content-Length: 0. >>> . >>> >>> >>> U 2011/12/01 22:51:11.559527 192.168.1.145:5090 -> 10.2.3.245:5060 >>> CANCEL i...i.. SIP/2.0. >>> Via: SIP/2.0/UDP 192.168.1.145:5090;branch=z9hG4bK5421.22999dd2.0. >>> B2B.256.3572553sip:+12125551212@10.2.3.245sip:888444@192.168.1.1379120d3`.p..i...q.i >>>  CANCEL. >>> User-Agent: OpenSIPS (1.7.1-notls (x86_64/linux)). >>> Max-Forwards: 70. >>> User-Agent: Opensips. >>> Init-CallID: 40c30c6459b3eaa4683991082381cadb@192.168.1.137. >>> Contact: . >>> . >>> >>> On Dec 07, 2011, at 05:18 PM, Ovidiu Sas  wrote: >>> >>> Add a log and print out what are you adding before adding it and you >>> will see if it's good or not. >>> >>> On Wed, Dec 7, 2011 at 5:13 PM, Logan  wrote:  This is the extent of my local route. If the $var is not present, I do not  add it. Do you see any issue with what I'm doing here?    local_route {          #xlog("L_INFO","* IN LOCAL ROUTE \n");           if (is_method("INVITE")) {                  if($var(pai_userpart)) {                          append_hf("P-Asserted-Identity:  \"$var(pai_display)\" \r\n");                  }else{                          xlog("L_INFO","PAI is not present, not adding\n");                  }          }    }   On Dec 07, 2011, at 04:57 PM, Ovidiu Sas  wrote:   You need to be careful when you alter requests in B2B mode (the  received INVITE and the sent INVITE belong to different transactions).  Make sure that you have something valid in those vars before applying  any changes to the outgoing message.   Regards,  Ovidiu Sas   On Wed, Dec 7, 2011 at 4:49 PM, Logan  wrote: > I'm storing some $vars in route[0] prior to calling b2b_init_request("top > hiding"); > > Then in my local route Im appending a P-Asserted-Identity header. > > I can't use the custom_headers modparam because it's going to preserve > the > P

[OpenSIPS-Users] SIP/SIMPLE to XMPP Gateway for SMS

[DMF]

Hi all, 

I'm looking for a SIP/SIMPLE to XMPP gateway solution and my google searches
have brought me here. I have an account with a voip provider that supports
SMS via SIP/SIMPLE MESSAGE (http://tools.ietf.org/html/rfc3428).
Unfortunately, I'm stuck using an old Blackberry which doesn't have any good
SIP apps, the only one that I've found that supports RFC3428 doesn't work
properly. I'm hoping that I can setup OpenSIPS to receive the SIP/SIMPLE
messages from my voip provider and forward them to an XMPP server like
openfire and be able to receive/respond to these messages via an XMPP
client. Would this be possible with OpenSIPS? Also, would it be possible to
have each message come from a unique XMPP user so that responses can be
tracked to the proper source? 

Thanks.

--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-SIMPLE-to-XMPP-Gateway-for-SMS-tp7097996p7097996.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] uac_replace_from corruption

[Jeff Pyle]

Brett,

Is the other end an Acme?  If so, they need to implement some custom
parameters (which I do not have) to* *honor some parts of section 12 of
RFC3261 in such a way that won't break uac_replace_from().  Let me know if
this is the case and we'll talk more.


Rasvan,

Can you share more about the "new" way to do it with the dialog module?  Is
this available in 1.6?



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

Re: [OpenSIPS-Users] Siptraces not shown on OpenSIPS-CP

[spady]

Can someone help me with this?

I checked again config and seems ok but form CP nothing yet.

Regards

--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/Siptraces-not-shown-on-OpenSIPS-CP-tp7052741p7097614.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] how does OpenSIPS manage 183's message??

[spady]

Hi Denis, that was it!!! It was setted to "auto" . 
I set it to "none" and now it works as aspected Perfet.
Thank you very much for your hint ;-)
Best regards

--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097238.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] how does OpenSIPS manage 183's message??

[dpa]

As I understand you use uac_replace_to() function to change TO header, and as I 
can see from debug you have "restore_mode" parameter 
(http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id249076) in UAC 
module which was established to value 3 (or not established at all). So 
Opensips restores TO URI to initial value.


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Thursday, December 15, 2011 5:32 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi Denis, I know. it's wanted.
Is changed to 87019.



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097102.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


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

[OpenSIPS-Users] Calls stuck in dialog state 5?

[Chris Martineau]

Hi,

 

I have noticed that I have a number of calls stuck in the dialog list in
state 5 (dialog ended)?

 

Apart from restarting the system I cannot see a way of clearing these.

 

I assume they should clear themselves and most do but for some reason
these are not.

 

I use database persistent dialogs but these only don't show in the
database, only in the cache?

 

Any ideas

 

Version 1.6.4

 

 

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

Re: [OpenSIPS-Users] Challenge authentication with sip gateway

[Ovidiu Sas]

You can use the b2b modules to perform authentication.
You will need to load uac_auth first and provision the credentials in
the credentials modparam:
http://www.opensips.org/html/docs/modules/1.7.x/uac_auth.html#id249026
You need to match the realm from 401 with the realm from modparam.


Regards,
Ovidiu Sas

--
VoIP Embedded, Inc.http://www.voipembedded.com

On Thu, Dec 15, 2011 at 8:24 AM, Arnold Vriezekolk NETZOZEKER B.V.
 wrote:
> Hi,
>
> I'm trying to achieve the following scenario:
>
> SIP Phone <-> OpenSIPS <-> SIP Provider
>
> From my SIP Phone i want to dial a number, let opensips do the challenge
> authentication to the sip provider and get the call connected.
>
> What i currently have in opensips is an extension '1234' which redirects to
> the sip gateway with a mobile number. Whenever i dial 1234 on my sip phone
> opensips relays the '401 unauthorized' message from the sip provider back to
> my phone, and the phone tries to authenticate itself with the sip provider.
>
> What i want it to do is let opensips handle the challenge authentication and
> connect the call back to my phone. I'm wondering if what i'm looking for is
> B2BUA. Could anyone give me a hint if i'm going in the right direction or
> that
> i maybe need to switch to the B2BUA module?
>
> Below you will find my opensips.cfg and the siplog of the call. In the
> siplog you will see my phone with the username HIJKLMNO.
>
> Thanks in advance!
>
> Arnold Vriezekolk

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

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi Denis, I know. it's wanted.
Is changed to 87019.



--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097102.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] Challenge authentication with sip gateway

[Arnold Vriezekolk NETZOZEKER B . V .]

Hi,

I'm trying to achieve the following scenario:

SIP Phone <-> OpenSIPS <-> SIP Provider


From my SIP Phone i want to dial a number, let opensips do the challenge

authentication to the sip provider and get the call connected.

What i currently have in opensips is an extension '1234' which redirects to
the sip gateway with a mobile number. Whenever i dial 1234 on my sip phone
opensips relays the '401 unauthorized' message from the sip provider back to
my phone, and the phone tries to authenticate itself with the sip provider.

What i want it to do is let opensips handle the challenge authentication and
connect the call back to my phone. I'm wondering if what i'm looking for is
B2BUA. Could anyone give me a hint if i'm going in the right direction or that
i maybe need to switch to the B2BUA module?

Below you will find my opensips.cfg and the siplog of the call. 
In the siplog you will see my phone with the username HIJKLMNO.


Thanks in advance!

Arnold Vriezekolk


[1] opensips.cfg: http://pastie.org/private/kmoobgua9okzwbnirhrq
[2] siplog: http://pastie.org/pastes/3020482/text?key=sxnaqdgkbekygjqwxukgiq
(Replaced all the usernames, passwords and ip's from the config and log.)

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

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[dpa]

Hello

Look at your TO header in INVITE message
Opensips changes TO header in INVITE.

-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Tuesday, December 13, 2011 8:13 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

Hi all, I am still testing my solution to provide some additional features to
own Fax server, thanks to OpenSIPS.

IP-PBX<> OpenSIPS <> Fax Server

I am using OpenSIPS in stateless mode ( so without record-route ) and this
is the sip trace at OpenSIPS level

U 2011/12/13 15:46:04.075195 172.16.52.7:5061 -> 10.9.101.166:5060
INVITE sip:0363394686180@10.9.101.166:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel ;tag=11D255FDC152.
To: sip:0363394686180@10.9.101.166:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Max-Forwards: 70.
Contact: .
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28.
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=XMedius-Fax-Gateway 55439616 616 IN IP4 172.16.52.7.
s=SIP Fax Call.
c=IN IP4 172.16.52.7.
t=0 0.
m=audio 46022 RTP/AVP 18 8.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=ptime:20.


U 2011/12/13 15:46:04.080484 10.9.101.166:5060 -> 172.16.52.7:5061
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel ;tag=11D255FDC152.
To: sip:0363394686180@10.9.101.166:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Server: OpenSIPS-Longwave.
Content-Length: 0.
.


U 2011/12/13 15:46:04.080927 10.9.101.166:5060 -> 10.9.6.3:5060
INVITE sip:87019363394686180@10.9.6.3:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
From: Fax Alcatel ;tag=11D255FDC152.
To: sip:87019363394686180@10.9.6.3:5060.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Max-Forwards: 69.
Contact: .
User-Agent: Alcatel-Lucent OmniTouch Fax Server Application/6.5.6.28.
P-Alcatel-CSBU: charging=sip:2542@ucalcatel.sedoc.locale.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=XMedius-Fax-Gateway 55439616 616 IN IP4 172.16.52.7.
s=SIP Fax Call.
c=IN IP4 172.16.52.7.
t=0 0.
m=audio 46022 RTP/AVP 18 8.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=ptime:20.


U 2011/12/13 15:46:04.082838 10.9.6.3:5060 -> 10.9.101.166:5060
SIP/2.0 100 Trying.
To: sip:87019363394686180@10.9.6.3:5060.
From: Fax Alcatel ;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 0.
.


U 2011/12/13 15:46:04.506636 10.9.6.3:5060 -> 10.9.101.166:5060
SIP/2.0 183 Session Progress.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:87019363394686180@10.9.6.3:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel ;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 10.9.101.166;branch=z9hG4bK8106.0a3d7b63.0.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 229.
.
v=0.
o=OXE 1323791155 1323791155 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.8.
t=0 0.
m=audio 32560 RTP/AVP 18 96.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.
a=rtpmap:96 telephone-event/8000.
a=sendrecv.


U 2011/12/13 15:46:04.508542 10.9.101.166:5060 -> 172.16.52.7:5061
SIP/2.0 183 Session Progress.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:0363394686180@10.9.101.166:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel ;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 101 INVITE.
Via: SIP/2.0/UDP 172.16.52.7:5061.
Content-Length: 229.
.
v=0.
o=OXE 1323791155 1323791155 IN IP4 10.9.6.3.
s=abs.
c=IN IP4 10.9.6.8.
t=0 0.
m=audio 32560 RTP/AVP 18 96.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=ptime:30.
a=maxptime:40.
a=rtpmap:96 telephone-event/8000.
a=sendrecv.


U 2011/12/13 15:46:06.028272 10.9.6.3:5060 -> 10.9.101.166:5060
SIP/2.0 180 Ringing.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS,
UPDATE.
Contact: sip:10.9.6.3.
User-Agent: OmniPCX Enterprise R9.0 h1.301.50.
P-Alcatel-CSBU: categparty=external.
Content-Type: application/sdp.
To:
sip:87019363394686180@10.9.6.3:5060;tag=3439ea69c02c4dd7146a60c535fa4a06.
From: Fax Alcatel ;tag=11D255FDC152.
Call-ID: 219ffbfb-58a0-41e1-acdc-82966f3a1f49@172.16.52.7.
CSeq: 

Re: [OpenSIPS-Users] how does OpenSIPS manage 183's message??

[spady]

Hi, Can someone explain/help me on this 
I really will appreciate 
Best Regards

--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/how-does-OpenSIPS-manage-183-s-message-tp7090462p7097041.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] uac_replace_from corruption

[Razvan Crainea]

Hi, Brett!

Are you using uac_replace_from function with dialog support or the old 
approach with route parameters? Would it be possible to increase 
OpenSIPS debugging level and paste some extra information?


Regards,

--
Ra(zvan Crainea
OpenSIPS Developer


On 12/15/2011 12:41 AM, Brett Nemeroff wrote:

Hello List,
I'm sure I brought this up sometime before, but I'm still having this 
issue. I have to imagine that some of you out there are having it as 
well. Running OpenSIP-1.6.4


I perform a uac_replace_from. It works just fine, But on subsequent 
transactions within the dialog, like a BYE that comes from the 
destination, the restored From is corrupted IF there was a port in the 
original From. This appears to be isolated to cases where the URI has 
a port in it.


For example, here's the From from the original INVITE
From: "8005551212" >;tag=as3371c611.


And then the far end sends a BYE later on:
From: "18705551313" >;tag=SDsjh0f99-eb402829


And I relay the BYE:
From: "18705551313" 
;tag=SDsjh0f99-eb402829


So aside from getting the client to stop sending the port in the From 
header, is there another fix for this?


Thanks!
-Brett



___
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