Re: [OpenSIPS-Users] Does OpenSIPS support RFC 5626 (Require Outbound)

[osiris123d]

Nevermind

Missed the part about the REGISTER request having a Loose Route

In message #9, Bob's UA sends its first registration through the
   first edge proxy in the outbound-proxy-set *by including a loose
   route*

*Route: *

If Snom can't do that then it cannot register twice I don't think.



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Re: [OpenSIPS-Users] CDRTool - Rating Origination and Termination differently

[osiris123d]

I fell off this for a while.  Did this ever get officially put into CDRTool?



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Re: [OpenSIPS-Users] Presence_XML issue with version 1.9

[osiris123d]

Nevermind.  I just saw that version 1.9 has a new XCAP module.  Someone might
need to update the modules documentation webpage to have a link to the XCAP
module documentation.  It's currently not there.

http://www.opensips.org/Resources/DocsModules19



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Re: [OpenSIPS-Users] Question about Parallel Forking

[osiris123d]

Also if you are expecting a second call to come in and want to send a BUSY if
there is already an ongoing call then you will want to look at the Dialog
modules "Exported Functions" to keep up with concurrent calls.  You might
want to look at them and figure out which one works best for your scenario
and you can do searches on them and see examples in the mailing list.



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Re: [OpenSIPS-Users] Question about Parallel Forking

[osiris123d]

Yes is this possible and probably has been discussed a lot on the mailing
list.  Please search on
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html

And also look at
http://www.opensips.org/Resources/DocsCoreFcn18#toc106
and
http://www.opensips.org/Resources/DocsCoreFcn18#toc144
and
http://www.opensips.org/Resources/DocsCoreVar18#toc20



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Re: [OpenSIPS-Users] Two OpenSIPS proxies issue

[osiris123d]

This is driving me crazy.  I was right the first time when I said that one of
the ACKs was not showing up as a loose route.  It is the third ACK that is
coming from the OpenSIPS/Proxy.  When it reaches the OpenSIPS/SBC device the
ACK fails as a loose route.

It would make sense that this would not be a loose route because there are
no Route headers so the loose_route() function would return FALSE.

The issue still remains that when the ACK reaches the OpenSIPS/SBC it still
isn't routed to the Callee, instead it is looped and routed to the same
interface it came from because that is whats in the RURI.

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Re: [OpenSIPS-Users] Persistent Usrloc records without using DB Backend

[osiris123d]

I read in the module documentation that it was for demo use.  I guess the
word demo made me think that it really shouldn't be used in production.  I
will try DB_Text out.

Thanks

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Re: [OpenSIPS-Users] Call pickup

[osiris123d]

Bogdan,

I'm trying to figure out how to get Call Pickup working since the PSTN
provider can't handle the Replaces: header.  Here is my post here

http://opensips-open-sip-server.1449251.n2.nabble.com/B2B-with-Call-Pickup-td7580224.html

I see in this post you talk about using MI commands and the TM and Dialog
modules and the failure route to make this work.  I think with the TM module
I can send a CANCEL to the original Callee but how would you make the call
then fail over to the Failure Route so I can send it to the next callee?

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[OpenSIPS-Users] B2B with Call Pickup

[osiris123d]

In the past I used a VoIP SIP trunk provider that I guess used Asterisk
servers as their gateways and I was able to perform the Call Pickup feature
with OpenSIPS and the SIP Trunk Provider. Now I have a new provider and I
need to use the B2B modules for Transfers.  Currently I am not able to
perform Call Pickup with calls coming from the PSTN to internal customers. 
I tried to use the b2b_bridge_request but when the PSTN Caller calls
internal_user_A and internal_user_B tries to do a call pickup I see the
following error in syslog

 ERROR:b2b_logic:b2bl_bridge_msg: Wrong state for entity ek= [B2B.106.390],
tk=[536.0]

When I do an "opensipsctl fifo b2b_list" all the states are zero.

So does the state of the call have to be a "on call" state instead of a
"calling" state?  How can you bridge a call that is in process of ringing?

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Re: [OpenSIPS-Users] B2BUA - Multiple Transfer/Refer issue

[osiris123d]

Issue is solved.  Anca Vamanu looked at the issue and said there was already
an answer for this issue.  Here is her answer

It seems that if you put the  node, your problems with reusage of id
will be solved. 

The scenario part will look like this:


client2
**

Refer-To



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Re: [OpenSIPS-Users] B2BUA - Multiple Transfer/Refer issue

[osiris123d]

Here are some Level 6 Debugs on the OpenSIPS B2BUA server.  


Good Transfer
http://pastebin.com/CcvUnN6y


Bad Transfer and a Transfer has already been done before
http://pastebin.com/j7BeEeJH


Does anyone else use the B2BUA model and the REFER script?  Are you able to
do multiple call transfers? 

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Re: [OpenSIPS-Users] Manipulate SIP clients "Allow" methods when registering

[osiris123d]

I am trying to edit the Contact Header for the Register message before it is
saved.  Here is what the contact header looks like

Contact:
;reg-id=1;q=1.0;audio;mobility="fixed";duplex="full";description="snom720";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,*MESSAGE*,INFO".

Any idea on how to use the subst() function in order to remove the MESSAGE
(and the comma too) part from the contact header??



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Re: [OpenSIPS-Users] Best way to Integrate Presence with IM in VoIP environment

[osiris123d]

Let me get some better examples on the Jitsi issue I noticed and post them.

As for the Bria issue if you manually enter the credentials it works but it
doesn't use the correct RFC path for the lists.

Example
Bria with credentials manually entered
GET /[hidden
email]/resource-lists/users/*9012XX2XX*/contacts-resource-list.xml HTTP/1.1. 

Bria without credentials manually entered
GET /[hidden
email]/resource-lists/users/*sip:9012xx...@irock.com*/resource-list.xml
HTTP/1.1.
Authorization: Digest username="*sip*", realm="irock.com",
nonce="50095226763860567326458028738337048298233814822458738089",
uri="/[hidden email]/resource-lists/users/sip:[hidden
email]/resource-list.xml"

You can see that with the manually entered example that just the username is
used in the path.  With the example where the SIP credentials are used you
see the full AOR used in the path.  The only problem is that for the
username the *sip:* is used from the AOR.

Anyway.  I'd rather not use Bria at this time with manual credentials
for fear that in the future things will change and the old xcap docs won't
be reachable because the path will be fixed.


But I will get you some examples on the Jitsi issue.  Thanks for the reply
back.

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Re: [OpenSIPS-Users] Can't get TLS working

[osiris123d]

Many hours wasted on this.

The issue


/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes


enough said.   (hitting head!)

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Re: [OpenSIPS-Users] New: Closeddial module

[osiris123d]

Ran into the REFER issue with Closeddial today.  Closeddial will not function
with REFER and B2B but I was still able to make it work by doing an
avp_db_query to grab the username of the cd_username and then did a
subst('/^Refer-To: sip:([0-9]+)@(.*)$/Refer-To:
sip:$avp(cdUsername)@blah.com./i');

So closeddial does and doesn't work with REFER :)

Depending on your OpenSIPS design it is possible to make it work.

Just thought I would update on this topic.

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Re: [OpenSIPS-Users] Does create_dialog behave differently on 1.6 than on 1.7?

[osiris123d]

Saul or Vlad,

Was this ever resolved?  I think I might have the same issue and I am using
the trunk version.

I have a Blink user behind NAT that registers and sends invites with private
IP in its VIA and Contact headers.  I do fix_nated_contact and
fix_nated_register without issue.  The only problem I have is when OpenSIPS
sends the OPTIONS message for Dialog keepalives.  It sends the OPTIONS to
the private IP address instead of the Public IP.  All the other messages in
the dialog get sent to the public IP and the call is established.  Also in
the Location table everything has the public IP for "contact" and
"received".

Here is the link to Pastebin for the ngrep and the location record for the
caller

http://pastebin.com/SMMsP03m

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Re: [OpenSIPS-Users] Caller doesn't hear ringing in ear when using append_branch and serial_branches/next_branches

[osiris123d]

The issue is that I don't have a media server. This is all done with
OpenSIPS relaying. Here is an example that kind of goes against what you
say with the 180 and 183 messages.

http://pastebin.com/NF6D6f4v

In this example the Caller first calls a user local to OpenSIPS. Then the
call is forked and a user out on the PSTN network is called. Finally
another fork occurs and a local user is called.

local user 9012732009 Q = 90 Called First Caller hears ringing
PSTN user 90121X8X63 Q = 50 Called Second Caller hears ringing
local user 9013349019 Q = 40 Called Last Caller hears ringing

So you would think that on the last call the Caller would not hear ringing
in his ear but he does.

So my first example had the following
Callee1 sends a 183
Callee2 sends a 180 < No ringing

My second example had the following
Callee1 sends a 180
Callee2 sends a 183
Callee3 sends a 180 < Caller can hear ring in ear just fine


So on my second scenario is it because the first callee sends a 180 that
the third callee sending a 180 doesn't mess things up??? Perhaps this
isn't a bug and we can move this conversation to the mailing list, but I
figure OpenSIPS would need a way to fix this or else other people will run
into this say issue when appending branches and calling out to PSTNs that
send back 180 when the second callee sends a 183.

I tested the first scenario with Blink being the Caller and Blink was not
able to hear ringing in the ear when the second callee was called.

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Re: [OpenSIPS-Users] Caller doesn't hear ringing in ear when using append_branch and serial_branches/next_branches

[osiris123d]

Anyone have any clue or guess what might cause this?

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Re: [OpenSIPS-Users] ICE and Mediaproxy break when using append_branch and serial_branches/next_branches

[osiris123d]

Saul,

I just realized that when the Bria client is the caller then I see the
one-way audio issues.  If the caller is a Snom phone then I don't see the
one-way audio issue.  Would it be possible to send you the SIP traces and
syslogs showing you the difference between a Snom call and a Bria call?  I
don't really see a difference that could cause this.



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Re: [OpenSIPS-Users] Trying to get Snom Contact List to work with OpenSIPS RLS

[osiris123d]

My main.c file has the following at the top
 
/*
 * $Id: main.c 8506 2011-10-21 10:21:10Z vladut-paiu $



So do you think there is any point to installing the latest trunk version
and seeing if it works or do you think the bug will still be in the latest
version?  Should I open a ticket on sourceforge?

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Re: [OpenSIPS-Users] Auth_DB's load_credentials isn't grabbing RPID info

[osiris123d]

I think I know the issue with this.  For INVITES I am using the
cachedb_local.  So if a user dials out the first time then credentials are
pulled from the database and the RPID info is gotten.  The second time the
user calls someone the cached credentials are used and the RPID is never
pulled.  So that is why my RPID is not getting attached to my INVITES.

Is there a way to fix this or should I just not use cachedb_local setup when
INVITES are being authenticated???

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Re: [OpenSIPS-Users] Localcache not working anymore

[osiris123d]

Please disregard.  This was my issue

I had the following parameters set up
modparam("auth_db", "load_credentials", "$avp(55)=ha1")
## The following is for RPID  
modparam("auth_db", "load_credentials", "$avp(rpid)=rpid")


Looks like I will need to combine into one parameter.

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Re: [OpenSIPS-Users] Restricting Max Contacts but allowing Softphone

[osiris123d]

The main reason that brought about this post is that for some reason my
devices are registering and for some reason I have multiple Location
entries.  Those extra entries seem to make weird things happen like multiple
dialogs when a call is made.  Or just extra overhead because multiple SIP
messages are being sent out.  I'm not sure why I am all of the sudden
getting multiple registration entries but it would be good if you could
limit the entries somehow other then by MAX number or IP.

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Re: [OpenSIPS-Users] ERROR:mediaproxy:__tm_request_in: could not create new dialog

[osiris123d]

I'm having the same issue and I haven't noticed these errors in the past.  I
am running on a Opensips trunk version that I updated last Saturday.  I am
manually executing create_dialog within my script but that is so I can do
the OPTION ping of caller, callee or both.  I can reproduce this every time. 
I will email the debug and a siptrace since the debug is too big to go on
pastebin.

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Re: [OpenSIPS-Users] ICE and Mediaproxy break when using append_branch and serial_branches/next_branches

[osiris123d]

Saul,

Looks like you've been pretty busy.  Just wanted to see if you ever had a
chance to mock this up?

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Re: [OpenSIPS-Users] serialize_branch breaking $du

[osiris123d]

bump :)

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Re: [OpenSIPS-Users] Q Value not acting right after Serialize_branches

[osiris123d]

bump :)

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Re: [OpenSIPS-Users] Dialog Ping

[osiris123d]

When the call is initially set up between the two clients should I see sip
OPTIONS messages being sent to both clients every X seconds?  I have this
set up and I don't see any OPTIONS messages being sent at all during the
duration of the good call.

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Re: [OpenSIPS-Users] Add Q Value to current SIP URI

[osiris123d]

http://www.opensips.org/Resources/DocsCoreVar#toc74

Works like a charm.

Thanks guys.

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Re: [OpenSIPS-Users] Timer based Failover to SIP Provider

[osiris123d]

I think I figured this out.  The Dispatcher module helped out and I think my
fr_timer and fr_inv_timer were giving me my issues.  After tons of
troubleshooting I think I have good failover for my SIP trunks.



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Re: [OpenSIPS-Users] Timer based Failover to SIP Provider

[osiris123d]

I still can't figure out how to fix this issue.  If the OpensipsB2BUA server
wasn't in the scenario the timeout failover would work.  Is it possible to
somehow generate a new CALL-ID for the second "failover" call?  Or do I need
to do something like catch the CANCEL on the OpensipsProxy and just send it
to Null or something???

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Re: [OpenSIPS-Users] CDRTool - MySQL Database error: Invalid SQL:

[osiris123d]

Figured out the issue but it doesn't make sense.  From my last post it
appeared that when mysql was querying for the radacct201106 table it was
looking in the cdrtool database.  Well when I set up cdrtool's global.inc
config I made it so that cdradmin mysql user was used to log into the
cdrtool database and also the radius database.  Here is an example of what I
had

class DB_CDRTool extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "cdrtool";
  var $User = "cdradmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_Locker extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "cdrtool";
  var $User = "locker";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_radius extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "radius";
  var $User = "cdradmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_opensips extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "opensips";
  var $User = "opensips";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_mediaproxy extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "mediaproxy";
  var $User = "cdradmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";



Originally I had the radius database user as "radius", but when I tried to
create that user in mysql it wouldn't let me (maybe because there is a
database called radius, not sure).  So I figured I would just use the
cdradmin user for everything except for opensips database.  Well when I
changed the cdrtool config to the following it is now about to do CDR
searches


class DB_CDRTool extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "cdrtool";
  var $User = "cdradmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_Locker extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "cdrtool";
  var $User = "locker";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_radius extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "radius";
  var $User = "radadmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_opensips extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "opensips";
  var $User = "opensips";
  var $Password = "*";
  var $Halt_On_Error ="yes";
}

class DB_mediaproxy extends DB_Sql {
  var $Host = "127.0.0.1";
  var $Database = "mediaproxy";
  var $User = "cdradmin";
  var $Password = "*";
  var $Halt_On_Error ="yes";


So it all appears to be good now, but I figured I should have still been
able to use the cdradmin user for both databases.  For some reason CDRTool
was looking in the cdrtool database for the radacct201106.



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Re: [OpenSIPS-Users] CDRTool - MySQL Database error: Invalid SQL:

[osiris123d]

I enabled logging on the MySQL server and when I click Search on the CDR
search and then go to the log after it fails with the "MySQL error: 1146
(Table 'cdrtool.radacct201106' doesn't exist) Session halted." error I see
the following in the log


110601 13:28:4413 Connect   cdradmin@localhost on 
   13 Init DB   cdrtool
   13 Query select val from active_sessions where sid  =
'47e6b96d5dbb0405bba6d9d5181f392a' and name = 'CDRc'
   13 Init DB   radius
   13 Query SELECT 1
   13 Init DB   cdrtool
   13 Query select * from sip_status order by code
   13 Query select `value` from memcache where `key` =
'destinations'
110601 13:28:4513 Query select `value` from memcache where `key` =
'destinations_sip'
   13 Query select * from billing_enum_tlds
   14 Connect   opensips@localhost on 
   14 Init DB   opensips
   14 Query select * from domain
   14 Query select * from trusted
   13 Query SHOW TABLES
   13 Query select count(*) as records
from radacct201106 where  (AcctStartTime >= '2011-06-01 12:18'
and AcctStartTime < '2011-06-01 23:55')



This makes me think that CDRTool is looking for the radacct201106 table in
the OpenSIPS database.  Am I wrong?

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[OpenSIPS-Users] CDRTool - MySQL Database error: Invalid SQL:

[osiris123d]

I have a fresh install of CDRTool on an Ubuntu box.  Everything appears to be
ok except for the fact that when I go to the CDR page and do a search I get
the following error on the webpage

MySQL error: 1146 (Table 'cdrtool.radacct201106' doesn't exist) Session
halted. 

Then when I look in syslog I see the following

Jun  1 16:35:57 CDRTool01 cdrtool[5986]: Database error: Invalid SQL: select
count(*) as records#012from radacct201106 where  (AcctStartTime
>= '
2011-06-01 10:25' and AcctStartTime < '2011-06-01 23:55') 
Jun  1 16:35:57 CDRTool01 cdrtool[5986]: 64

Yet if I log into mysql and perform the same command (minus the #12 junk
that is in syslog) it returns a value without issue.

Also in syslog I see other MySQL database errors



Jun  1 16:35:17 CDRTool01 cdrtool[21095]: Database error: Invalid SQL:
select *, UNIX_TIMESTAMP(AcctStartTime) as timestamp#012from
radacct201106 whe
re  (1=1)  and  Normalized = '0' and AcctStopTime not like '-00-00
00:00:00%'  order by RadAcctId asc
Jun  1 16:35:17 CDRTool01 cdrtool[21095]: 64



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Re: [OpenSIPS-Users] CDRTool install

[osiris123d]

Did ram ever respond to this question?

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Re: [OpenSIPS-Users] Timer based Failover to SIP Provider

[osiris123d]

The only reason why I think the second call is connecting and then hanging up
is because the second call has the exact same Call-ID as the first call.  So
when the OpenSIPSProxy sends the CANCEL for the first call to the
OpenSIPSB2BUA the second call has already started connecting and sending the
183 to the OpenSIPSProxy.  So the second call has just enough time to ring
the remote phone as OpenSIPSProxy and OpenSIPSB2BUA are tearing down the
initial call which has the exact same Call-ID.  Doesn this make any sense?  

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Re: [OpenSIPS-Users] Reroute B2B call after failure

[osiris123d]

Do you have some code example of how to do this?  I am not sure what
variables need to be overwritten.

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Re: [OpenSIPS-Users] BLA/SLA not working

[osiris123d]

Hey Anca,

It has been a while but I just noticed that the Snom people replied to my
post about this issue but I didn't get a notice about their post.  Here is
what they said


Hello,

I don't think this is the problem. The phone starts sending the 'direction'
tag as soon as the dialog is initialized (the second NOTIFY in your trace).
But for some reason the server still fails to parse it:

Nov 8 21:13:56 Proxy01 /usr/local/sbin/opensips[15517]:
ERROR:presence:bla_aggregate_state: Dialog direction not specified <--
the server says the direction tag is not present, but:
Nov 8 21:13:56 Proxy01 /usr/local/sbin/opensips[15517]:
INFO:presence:bla_aggregate_state: new:#012#012http://opensips-open-sip-server.1449251.n2.nabble.com/BLA-SLA-not-working-tp5739308p6373668.html
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Re: [OpenSIPS-Users] Killing confirmed no-ack dialogs

[osiris123d]

I just tried to set up my script this way and it is not working.  I have set
it up so that during the INVITE the timeout_avp gets set to 120 seconds. 
Then when the message is an ACK I set the timeout_avp to 7200 seconds.  When
I make a call to test the call will only stay up for 2 minutes.  I know that
my second declaration of timeout_avp is being executed since I did the
following

modparam("dialog", "timeout_avp", "$avp(i:30)")
modparam("dialog", "bye_on_timeout_flag", 30) 

if (has_totag()) {
  if (loose_route()) {

if ( is_method("ACK") ) {
# The dialog has reached 4th step in setting
up the call
# so everything should be good.  We want to
go ahead and
# set the dialog timeout to something high
now
xlog("L_INFO", "Main Route Just set Timeout to 7200: Call [$rm] rU[$rU]
fU[$fU]\n");
$avp(i:30)=7200;
setflag(30);
}

  }
}


Am I missing something here?

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[OpenSIPS-Users] Virtual_DB issue with NDBClUSTER

[osiris123d]

I am trying to get OpenSIPS Virtual_DB module to work with a MySQL Cluster
that I have set up.

Instead of the database tables being the usual MyISAM I set them up to be
NDBCLUSTER.

Within the OpenSIPS config I have

# - db_virtual params -
modparam("db_virtual", "db_urls", "define set1 ROUND") 
modparam("db_virtual", "db_urls",
"mysql://:***...@173.***.***.219/opensips")
modparam("db_virtual", "db_urls",
"mysql://:***...@173.***.***.218/opensips")


To test I shut down the database on the 173.***.***.218 server and OpenSIPS
went down.  When I try to restart OpenSIPS I see the following in the syslog

Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:db_mysql:db_mysql_connect: driver error(2003): Can't connect to MySQL
server on '173.***.***.218' (111)
Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:db_mysql:db_mysql_new_connection: initial connect failed
Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:core:db_do_init: could not add connection to the pool
Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:db_virtual:db_virtual_init: cant init db
mysql://:**...@173.***.***.218/opensips
Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:db_mysql:db_mysql_submit_query: driver error: Got error 157 'Unknown
error code' from NDBCLUSTER
Dec 14 11:17:20 Proxy01 /usr/local/sbin/opensips[3843]:
ERROR:core:db_do_query: error while submitting query
Dec 14 11:17:20 Proxy01 kernel: [235305.342266] opensips[3843]: segfault at
20 ip 004f4e03 sp 7fffb3965840 error 4 in
opensips[40+14d000]



Looks like it dumped the core and the core says

(gdb) backtrace
#0  0x004f4e03 in db_table_version (dbf=0x78d6e470,
connection=0x8175e0, table=0x7f3e124e9b40) at db/db.c:359
#1  0x004f51d1 in db_check_table_version (dbf=0x0, dbh=0x0,
table=0x7f3e124e9b40, version=7) at db/db.c:398
#2  0x7f3e122e585e in mod_init () at uri_mod.c:265
#3  0x00493514 in init_mod (m=0x798348) at sr_module.c:457
#4  0x004934ac in init_mod (m=0x798418) at sr_module.c:452
#5  0x004934ac in init_mod (m=0x7984e8) at sr_module.c:452
#6  0x004934ac in init_mod (m=0x7985b8) at sr_module.c:452
#7  0x004934ac in init_mod (m=0x798688) at sr_module.c:452
#8  0x004934ac in init_mod (m=0x798758) at sr_module.c:452
#9  0x004934ac in init_mod (m=0x798828) at sr_module.c:452
#10 0x004934ac in init_mod (m=0x7988f8) at sr_module.c:452
#11 0x004934ac in init_mod (m=0x7989c8) at sr_module.c:452
#12 0x004934ac in init_mod (m=0x798a98) at sr_module.c:452
#13 0x004934ac in init_mod (m=0x798b68) at sr_module.c:452
#14 0x004934ac in init_mod (m=0x798c38) at sr_module.c:452
#15 0x004934ac in init_mod (m=0x798d08) at sr_module.c:452
#16 0x004934ac in init_mod (m=0x798dd8) at sr_module.c:452
#17 0x004934ac in init_mod (m=0x798ea8) at sr_module.c:452
#18 0x004934ac in init_mod (m=0x798f78) at sr_module.c:452
#19 0x004934ac in init_mod (m=0x799048) at sr_module.c:452
#20 0x004934ac in init_mod (m=0x799118) at sr_module.c:452
#21 0x004934ac in init_mod (m=0x7991e8) at sr_module.c:452
#22 0x004934ac in init_mod (m=0x799528) at sr_module.c:452
#23 0x004934ac in init_mod (m=0x7996c8) at sr_module.c:452
#24 0x004934ac in init_mod (m=0x799798) at sr_module.c:452
#25 0x004934ac in init_mod (m=0x799868) at sr_module.c:452
#26 0x004934ac in init_mod (m=0x799a70) at sr_module.c:452
#27 0x004934ac in init_mod (m=0x799b40) at sr_module.c:452
#28 0x004934ac in init_mod (m=0x799c10) at sr_module.c:452
#29 0x004934ac in init_mod (m=0x799ce0) at sr_module.c:452
#30 0x004934ac in init_mod (m=0x799db0) at sr_module.c:452
#31 0x004934ac in init_mod (m=0x799e80) at sr_module.c:452
#32 0x004934ac in init_mod (m=0x799f50) at sr_module.c:452
#33 0x004934ac in init_mod (m=0x79a020) at sr_module.c:452
#34 0x0042c4f0 in main (argc=, argv=) at main.c:1351


Can you not use NDBCluster with OpenSIPS???
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Re: [OpenSIPS-Users] db_mysql core dump

[osiris123d]

I am still having the same core dumps every day.  I think it has something to
do with my Presence config in my script.  I am running the latest stable
version of OpenSIPS 1.6.3 from the download page.  I have reinstalled but
that doesn't help.  My Debian version is 2.6.35.4-rscloud since I run it on
a rackspace VM server.

More info from the backtrace shows the following

(gdb) frame 8
#8  main (argc=, argv=) at
main.c:1388
1388ret=main_loop();
(gdb) frame 7
#7  0x0042ce8b in main_loop (argc=, argv=) at main.c:867
867 if (start_timer_processes()!=0) {
(gdb) frame 6
#6  start_timer_processes () at timer.c:475
475 run_timer_process( tpl );
(gdb) frame 5
#5  run_timer_process () at timer.c:395
395 timer_ticker(tpl->timer_list);
(gdb) frame 4
#4  0x004a90b6 in timer_ticker () at timer.c:325
325 t->u.timer_f(*jiffies, t->t_param);
(gdb) frame 3
#3  0x7f176c25ba7a in msg_watchers_clean (ticks=,
param=) at subscribe.c:484
484 if (pa_dbf.delete(pa_db, db_keys, db_ops, db_vals, 2) < 0) 
(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) frame 1
#1  0x7f176faf6b06 in db_mysql_do_prepared_query (conn=0x812f20,
v=0x7fff61728900, n=2, uv=, un=,
query=) at dbase.c:443
443 if (db_mysql_val2bind( v+i , mysql_bind, i)<0 ) {
(gdb) frame 0
#0  0x7f176fafca93 in db_mysql_val2bind (v=0x7fff61728920,
binds=0x813690, i=) at val.c:274
274 *(binds[i].is_null) = 0;


So am I leaving some value in my OpenSIPS config NULL and this is causing
this segfault?  A lot of the time my OpenSIPS process core dumps during the
night when nothing is being used (its a test box).  So really the only thing
that should be going on are phones re-registering and presence requests. 
And just from trying to look through the .c files listed in the backtrace it
has something to do with cleaning up watcher subscriptions.

If anyone has any info on where I could look it would be helpful.  


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[OpenSIPS-Users] osipsconsole doesn't seem to work for me

[osiris123d]

Since osipsconsole came out I have never really used it because I can't get
it to work.  Now I am starting to look into why I can't get it to work.  I
followed the install guide but I am not sure what's wrong

Proxy01:/etc/opensips-mi-proxy# osipsconsole 
Control engine FIFO loaded
OpenSIPS$:domain show
Used database is mysql
coolbeans.com
 irock.com
 
OpenSIPS$:db show location
Unknown command
OpenSIPS$:

OpenSIPS$:add 9013319...@irock.com password1234
Used database is mysql
9013319320 irock.com
Entry could not be retrieved from tableYou have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the right
syntax to use near 'WHERE username = '9013319320' 
   AND domain = 'irock.com'' at line 3
Query results:





Here is the osicpsconsolerc file I configured (does it matter if the
database was originally created with opensipsctl??)

# $Id: osicpsconsolerc 
#
# The OpenSIPS configuration file for the control tools.
#
# Here you can set variables used in the opensipsctl and opensipsdbctl setup
# scripts. Per default all variables here are commented out, the control
tools
# will use their internal default values.

## your SIP domain
SIP_DOMAIN=ae.com

## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, or DBTEXT, by default
none is loaded
# If you want to setup a database with opensipsdbctl, you must at least
specify
# this parameter.
DBENGINE=MYSQL

## database host
DBHOST=173.XXX.XXX.219

## database port (PostgreSQL=5433 mandatory; MYSQL=3306 optional)
DBPORT=3306
#DBPORT=5000

## database name (for ORACLE this is TNS name)
DBNAME=opensips

## database path used by dbtext or db_berkeley
# DB_PATH="/usr/local/etc/opensips/dbtext"

## database read/write user
DBRWUSER=opensips

## password for database read/write user
DBRWPW=

## database read only user
# DBROUSER=opensipsro

## password for database read only user
# DBROPW=opensipsro

## database super user (for ORACLE this is 'scheme-creator' user)
DBROOTUSER=opensips

# Program to calculate a message-digest fingerprint 
# MD5=md5sum

# awk tool
# AWK=awk

# grep tool
# GREP=egrep

# sed tool
# SED=sed


# Describe what additional tables to install. Valid values for the variables
# below are yes/no/ask. With ask (default) it will interactively ask the
user
# for an answer, while yes/no allow for automated, unassisted installs.
#

# Define what module tables should be installed.
# If you use the postgres database and want to change the installed tables,
then you
# must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly
in the
# opensipsdbctl.base script.

# opensips standard modules
# STANDARD_MODULES="standard acc lcr domain group permissions registrar
usrloc msilo
#   alias_db uri_db speeddial avpops auth_db pdt dialog
dispatcher
#   dialplan drouting nathelper load_balancer"

# opensips extra modules
# EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist"


## type of aliases used: DB - database aliases; UL - usrloc aliases
## - default: none
# ALIASES_TYPE=DB

## MI_CONNECTOR control engine: FIFO, UNIXSOCK, UDP, XMLRPC
MI_CONNECTOR=FIFO:/var/run/opensips/fifo
#MI_CONNECTOR=UNIXSOCK:/var/run/opensips/socket
# MI_CONNECTOR=UDP:192.168.2.133:8000
#MI_CONNECTOR=XMLRPC:173.XXX.XXX.134:8080

## check ACL names; default on (1); off (0)
# VERIFY_ACL=1

## ACL names - if VERIFY_ACL is set, only the ACL names from below list
## are accepted
# ACL_GROUPS="local ld int voicemail free-pstn"

## do (1) or don't (0) store plaintext passwords
## in the subscriber table - default '1'
STORE_PLAINTEXT_PW=0

## OPENSIPS START Options
## PID file path - default is: /var/run/opensips.pid
PID_FILE=/var/run/opensips/opensips.pid

## OUTPUT control - default output is to SYSLOG
## 0=output to console, 1=output to syslog
SYSLOG=0

## Extra start options - default is: not set
# example: start opensips with 64MB share memory: STARTOPTIONS="-m 64"
# STARTOPTIONS=
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Re: [OpenSIPS-Users] Asterisk Integration - Manipulate Asterisk Contexts

[osiris123d]

Thanks for the suggestion Mike.
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[OpenSIPS-Users] B2B issues with To Header (I think)

[osiris123d]

I am playing with the B2B module and not having a lot of luck.  I am using my
original script and adding in the b2b_init_request.  I execute all of my
logic like lookup("location") so that the callee info can be set up
correctly.  After all of that I do the following

if(is_method("INVITE") && !has_totag()) {
b2b_init_request("refer");
exit;
}

This sends the following request to the callee phone
INVITE sip:9012732...@75.xxx.xxx.158:2074 SIP/2.0
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0
To: sip:9012732...@75.xxx.xxx.158:2074
From: ;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
CSeq: 3 INVITE
Call-ID: B2B.114.3927076
Content-Length: 451
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Type: application/sdp
Supported: timer, 100rel, replaces, from-change
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Session-Expires: 3600;refresher=uas
Min-SE: 90
Contact: 

v=0
o=root 535295098 535295098 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 65214 RTP/AVP 9 8 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:et2a2zK91Vh8Hk1o415DWp/kM1BtwbOTmJONkV9E
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv




Sent to udp:173.XXX.XXX.134:5060 at 23/12/2001 18:15:15:695 (482 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0
From: ;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
To: 
Call-ID: B2B.114.3927076
CSeq: 3 INVITE
User-Agent: snom360/8.4.18
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0



Because the TO header doesn't have the real domain on it the phone rejects
it

So I thought by using OpenSIPS local_route I could do the following
local_route {
if (is_method("INVITE")) {
remove_hf("To");
append_hf("To: \r\n");   
}
}



This doesn't seem to make a difference at all.  The callee phone still
rejects this.  here is what the phone does when I use local_route


INVITE sip:9012732...@75.xxx.xxx.158:1850 SIP/2.0
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0
From: ;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
CSeq: 3 INVITE
Call-ID: B2B.464.6147243
Content-Length: 451
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Type: application/sdp
Supported: timer, 100rel, replaces, from-change
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Session-Expires: 3600;refresher=uas
Min-SE: 90
Contact: 
To: 

v=0
o=root 808120215 808120215 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 64810 RTP/AVP 9 8 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:DXf894oyUu9RbqKk5DGs0bJtaJMlb9zi09qM4S7a
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv





Sent to udp:173.XXX.XXX.134:5060 at 23/12/2001 18:05:14:063 (480 bytes):

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0
From: ;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
To: 
Call-ID: B2B.464.6147243
CSeq: 3 INVITE
User-Agent: snom870/8.4.18
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0






Just to be sure I looked an Invite for a call that is good and successful.  

INVITE sip:9012732...@75.xxx.xxx.158:3072;line=hbpetirz SIP/2.0
Record-Route:

Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK0dbb.5dfc74b4.0
Via: SIP/2.0/UDP
192.168.33.23:2048;received=75.XXX.XXX.158;branch=z9hG4bK-97gss0xcllrx;rport=2048
From: "Moo 221-1612" ;tag=94usbbkjqi
To: 
Call-ID: 3c268edc0da6-3ut9py151hv1
CSeq: 2 INVITE
Max-Forwards: 69
Contact: ;reg-id=1
X-Serialnumber: 0004132902C9
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 453
P-hint: route(3)|setflag7,forcerport,fix_contact
P-hint: inbound->inbound 

v=0
o=root 1995837061 1995837061 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 54868 RTP/AVP 9 8 99 3 18 4 101
a=cry

Re: [OpenSIPS-Users] Asterisk Integration - Manipulate Asterisk Contexts

[osiris123d]

I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
address of callers phone as it appears in the location table.

On a side note I was able to not use P-Asserted-Identity.  because of a
different issue I learned about the uac_replace_to() function.  I was able
to place the real domain in the TO header.  Now with Asterisk I do the
following

exten => _VMS_.,1,Ringing
exten => _VMS_.,n,Wait(1)
exten => _VMS_.,n,Answer
exten => _VMS_.,n,Wait(1)
exten => _VMS_.,n,Set(dm=${SIP_HEADER(TO):16})
exten => _VMS_.,n,Set(dm=${CUT(dm,>,1)})
exten => _VMS_.,n,Voicemail(${EXTEN:4...@${dm},u)
exten => _VMS_.,n,Hangup

The offsets I use above assumes the username of the extension being called
is a 10 digit users NPANXX

Thanks for the reply.  The uac_replace_to() fixed multiple issues.


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[OpenSIPS-Users] Asterisk Integration - Manipulate Asterisk Contexts

[osiris123d]

I have set up Asterisk to work with OpenSIPS so that instead of the context
for all OpenSIPS Subscribers being "default" it is their actual domain. 
Following the OpenSIPS Tutorial to integrate with Asterisk worked well with
this until it came time for a x...@domaina to call a a...@domainb. 
x...@domaina gets sent to a...@domainb's voicemail if they don't pick up. 
Well with multiple contexts set up with Asterisk x...@domaina is going to be
funnelled into Asterisk context DomainA since he is apart of Context
DomainA.  This isn't good since context DomainA doesn't have a user called
ABC

I found a way around this

With OpenSIPS I do the following
if($rd != $var(callee_domain)){
prefix("VMS_"); 
append_hf("P-Asserted-Identity: $var(callee_domain)\r\n"); 
} 
else{
prefix("VMR_");
} 
rewritehostport("ASTERISK_IP");


And in Asterisk I did this

; Leave Voicemail for DomainA.com employee from External DID
exten => _VMR_.,1,Ringing
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Answer
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,Voicemail(${EXTEN:4...@coolbeans.com,u)
exten => _VMR_.,n,Hangup

; Leave Voicemail for other customer when DomainA.com employee calls them
exten => _VMS_.,1,Ringing 
exten => _VMS_.,n,Wait(1) 
exten => _VMS_.,n,Answer 
exten => _VMS_.,n,Wait(1) 
exten => _VMS_.,n,NoOp(${SIP_HEADER(P-Asserted-Identity)})
exten => _VMS_.,n,Voicemail(${EXTEN:4...@${sip_header(P-Asserted-Identity)},u)
exten => _VMS_.,n,Hangup 






This works!!  But what I want to know is am I breaking some rules here
by appending a P-Asserted-Identity SIP header and then looking for it in
Asterisk  

Comments???
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Re: [OpenSIPS-Users] rewritehostport not working, but it has before

[osiris123d]

I am doing a location() before that.  I guess before I had some logic set up
to where if the user wasn't in the location table then I would call the
rewritehost so that it would go to voicemail since the user wasn't logged
in.  Thats why it confused me because it worked just fine for that scenario. 
I will have to remember to watch out for $du.


Thanks


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Re: [OpenSIPS-Users] 2 UAs behind same NAT Device

[osiris123d]

If your phone registers with the OpenSIPS proxy and has it's Public IP
address show up in the location table then it is possible to see that two
devices are on the same LAN even though they might be in different subnets. 
You would just need to know when deploying phones to keep in mind if there
are any firewalls between devices.  You could also set up an AVPOP for every
subscriber, say LANSite, and it PhoneA(LANSite) == PhoneB(LANSite) then they
both can talk directly to each other without MediaProxy.  The only issue
with this is if the user moves their phone (say they take it home).  Thats
why you would need multiple IF statements to decide if the phones need
mediaproxy.  It's possible but it depends on your specific situation.
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Re: [OpenSIPS-Users] Asterisk Authentication with OpenSIPS integration

[osiris123d]

Thanks Flavio. That sounds like a good solution.  I will implement that.
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Re: [OpenSIPS-Users] rewritehostport not working, but it has before

[osiris123d]

revert_uri();
if (uri=~"^sip:9[2-9][0-9]{6}@") {
strip(1);
};
   
subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
prefix("VMR_");
rewritehost("173.203.78.63");
xlog("- new ruri=<$ru>, dst=<$du>\n");
$du= NULL;
xlog("- new ruri=<$ru>, dst=<$du>\n");
t_relay();  
exit;



Nov  3 15:49:17 Proxy01 /usr/local/sbin/opensips[25791]: Call control: user
channel limit exceeded [1/1]
Nov  3 15:49:17 Proxy01 /usr/local/sbin/opensips[25791]: - new
ruri=,
dst=
Nov  3 15:49:17 Proxy01 /usr/local/sbin/opensips[25791]: - new
ruri=, dst=<>


Doing the $du=NULL; fixes it.  But it shouldn't be doing this right?  In
other parts of my script I don't have to do this.
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Re: [OpenSIPS-Users] rewritehostport not working, but it has before

[osiris123d]

 revert_uri();
if (uri=~"^sip:9[2-9][0-9]{6}@") {
strip(1);
};
   
subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
prefix("VMR_");
rewritehost("173.XXX.XX.63");
xlog("- new ruri=<$ru>, dst=<$du>\n");

t_relay();
exit;


And my log shows

- new ruri=,
dst=
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[OpenSIPS-Users] rewritehostport not working, but it has before

[osiris123d]

I have used this in the past with the exact same route logic and it has
worked.  I am confused as to why it's not working now.  I am testing
limiting concurrent calls with the dialog module and have it set up to
reroute the call to Voicemail if the callee is over their limit of
concurrent calls.  Basically all I am doing is

# user has max channel limit set as preference
if(is_avp_set("$avp(s:channels)/n") &&
avp_check("$avp(s:channels)", "gt/i:0"))
{
# get current calls for uuid
   
get_profile_size("ConcurrentCalls","$rU","$var(CalleeCalls)");

# check within limit
if($avp(s:channels) > $var(CalleeCalls))
{
xlog("L_INFO", "Call control: user '$rU'
currently has '$var(CalleeCalls)' of '$avp(s:channels)' active calls before
this one\n");
$var(setprofileCallee) = 1;
}
else
{
xlog("L_INFO", "Call control: user channel
limit exceeded [$var(CalleeCalls)/$avp(s:channels)]\n");
#send_reply("486", "Busy Here: Channel limit
exceeded\n");

  revert_uri();
if (uri=~"^sip:9[2-9][0-9]{6}@") {
strip(1);
};
   
subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(s:areacode)\...@\2/i');
prefix("VMR_");
rewritehostport("173.XXX.XX.63");
t_relay();
exit;


}
}


So I call the 9012211610 who is only allowed one call with 9012732009.  Then
User 9012211612 also calls 9012211610.  OpenSIPS sees that 9012211610 is at
his max calls so OpenSIPS does the else statement listed above.  But the
INVITE isn't sent to 173.XXX.XX.63.  It is sent to the IP address of
9012211610.  Why is it doing this

U 2010/11/02 18:45:24.898279 75.65.8.158:61980 -> 173.XXX.XX.134:5060
INVITE sip:9012211...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.33.20:61980;branch=z9hG4bK-d8754z-933b5a7d10948cc3-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: "9012211610".
From: "Test Guy";tag=cba13467.
Call-ID: NzJlMGJmMGY1NDdjNzlmMmExYzdiNzI4ODEwZTY4MWU..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite 4 release 4.0 stamp 58832.
Content-Length: 232.
.
v=0.
o=- 12933215112580778 1 IN IP4 192.168.33.20.
s=CounterPath X-Lite 4.0.
c=IN IP4 192.168.33.20.
t=0 0.
m=audio 65100 RTP/AVP 107 0 8 101.
a=rtpmap:107 BV32/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.

#
U 2010/11/02 18:45:24.899628 173.XXX.XX.134:5060 -> 75.65.8.158:61980
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.33.20:61980;branch=z9hG4bK-d8754z-933b5a7d10948cc3-1---d8754z-;rport=61980;received=75.65.8.158.
To:
"9012211610";tag=c97b4d1cb1f3d0da549e06a8d482ef63.6250.
From: "Test Guy";tag=cba13467.
Call-ID: NzJlMGJmMGY1NDdjNzlmMmExYzdiNzI4ODEwZTY4MWU..
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="irock.com",
nonce="4cd0a2b20742bc39459aacb588bf623462cd9301", qop="auth".
Server: OpenSIPS (1.6.3-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 2010/11/02 18:45:24.955873 75.65.8.158:61980 -> 173.XXX.XX.134:5060
ACK sip:9012211...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.33.20:61980;branch=z9hG4bK-d8754z-933b5a7d10948cc3-1---d8754z-;rport.
Max-Forwards: 70.
To:
"9012211610";tag=c97b4d1cb1f3d0da549e06a8d482ef63.6250.
From: "Test Guy";tag=cba13467.
Call-ID: NzJlMGJmMGY1NDdjNzlmMmExYzdiNzI4ODEwZTY4MWU..
CSeq: 1 ACK.
Content-Length: 0.
.

#
U 2010/11/02 18:45:24.966584 75.65.8.158:61980 -> 173.XXX.XX.134:5060
INVITE sip:9012211...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.33.20:61980;branch=z9hG4bK-d8754z-be8a338c8dfe786b-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: "9012211610".
From: "Test Guy";tag=cba13467.
Call-ID: NzJlMGJmMGY1NDdjNzlmMmExYzdiNzI4ODEwZTY4MWU..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
Proxy-Authorization: Digest
username="9012732009",realm="irock.com",nonce="4cd0a2b20742bc39459aacb588bf623462cd9301",uri="sip:9012211...@irock.com",response="972e1c21b89d8582da0d399c59555cde",cnonce="d692791ba7ce1b1b221592550638ea88",nc=0001,qop=auth,algorithm=MD5.
Supported: replaces.
User-Agent: X-Lite 4 release 4.0 stamp 58832.
Content-Length: 232.
.
v=0.
o=- 12933215112580778 1 IN IP4 192.168.33.20.
s=CounterPath X-Lite 4.0.
c=IN IP4 192.168.33.20.
t=0 0.
m=audio 65100 RTP/AVP 107 0 8 101.
a=rtpmap:107 BV32/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=se

Re: [OpenSIPS-Users] Asterisk Authentication with OpenSIPS integration

[osiris123d]

Ok I got it work work with the following edit when creating the VIEW for
sipusers in the asterisk database

  `opensips`.`subscriber`.`password` AS `secret`,


But what I don't get is that OpenSIPS is using ha1 to verify passwords. 
This makes me think that Asterisk is reading from the password column in the
subscriber table of OpenSIPS.  If I try to do the following

  `opensips`.`subscriber`.`ha1` AS `md5secret`,

that messes it all up.  Any ideas?
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[OpenSIPS-Users] Asterisk Authentication with OpenSIPS integration

[osiris123d]

I am trying to get Call Queuing working with OpenSIPS and Asterisk.  I have
my Asterisk database set up just like the OpenSIPS&Asterisk Integration
tuturial.  When someone calls the Hunt Group number OpenSIPs forwards the
call to Asterisk.  Asterisk see's that the call needs to be put in a Queue
and does.  Then it sends out invites to OpenSIPS for the agents.  In
Asterisk I am seeing 
 chan_sip.c:19013 handle_response_invite: Failed to authenticate on INVITE
to 

I am pretty positive that Asterisk is not sending the right nonce info, but
I am not sure how to test it.  The default MySQL VIEW setup has secret set
to NULL.  I tried setting up secret to point to the password in the opensips
database and also set up MD5Secret to point to ha1.  But that doesn't seem
to help.


Anyone got any ideas?


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Re: [OpenSIPS-Users] B2BUA Implementation for Transfer

[osiris123d]

I take that back.  It is still Core dumping.  The core dump occurs 

Internal user calls Internal user
Callee Phone rings
Callee picks up call  <--- Core Dump occurs when callee picks up

(gdb) backtrace
#0  get_source_uri (dlg=0x7f2d91e48758, type=-1728747624,
_params=0x7f2d955b36e0) at nat_traversal.c:968
#1  __dialog_confirmed (dlg=0x7f2d91e48758, type=-1728747624,
_params=0x7f2d955b36e0) at nat_traversal.c:1106
#2  0x7f2d9538c7c2 in run_dlg_callbacks (type=8, dlg=0x7f2d91e48758,
msg=, dir=28514096, dlg_data=0x1b316d0) at dlg_cb.c:253
#3  0x7f2d95398a38 in dlg_onreply (t=0x7f2d91e46f00, type=, param=) at dlg_handlers.c:407
#4  0x7f2d976c23b9 in run_trans_callbacks (type=128,
trans=0x7f2d91e46f00, req=0x7f2d91e48878, rpl=0x, code=200)
at t_hooks.c:208
#5  0x7f2d976d3b0a in _reply_light (trans=0x7f2d91e46f00, 
buf=0x7d51d8 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP
192.168.33.22:2048;branch=z9hG4bK-hjarcoennd18;rport=2048;received=75.65.8.158\r\nFrom:
\"Poo 2211610\" ;tag=73uoyzaqch\r\nTo:
,
to_tag_len=, lock=1, bm=0x7fff4bb816d0) at
t_reply.c:385
#6  0x7f2d976d3d5e in t_reply_with_body (trans=0x7f2d91e46f00, code=200,
text=0x7d2db8, body=, new_header=,
to_tag=0x7f2d91e4a8f0)
at t_reply.c:1608
#7  0x7f2d9450dac7 in b2b_send_reply (et=,
b2b_key=0x7f2d91e4a8f0, code=200, text=0x7d2db8, body=0x7fff4bb81c80,
extra_headers=0x7fff4bb81c70, 
dlginfo=0x7f2d91e4a9a8) at dlg.c:1067
#8  0x7f2d942f52b4 in b2b_logic_notify (src=1, msg=, key=, type=1, param=) at
logic.c:768
#9  0x7f2d94510c9f in b2b_tm_cback (t=,
htable=0x7f2d91e305a8, ps=0xf) at dlg.c:1921
#10 0x7f2d976c23b9 in run_trans_callbacks (type=512,
trans=0x7f2d91e4ac30, req=0x0, rpl=0x7d2d88, code=200) at t_hooks.c:208
#11 0x7f2d976d59bf in local_reply (t=0x7f2d91e4ac30, p_msg=0x7d2d88,
branch=80192, msg_status=,
cancel_bitmap=0x7fff4bb82118) at t_reply.c:1340
#12 0x7f2d976d712f in reply_received (p_msg=0x7d2d88) at t_reply.c:1485
#13 0x0042514d in forward_reply (msg=0x7d2d88) at forward.c:559
#14 0x004664ed in receive_msg (
buf=0x761f20 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP
173.203.87.134;branch=z9hG4bKc229.ac1fd7d3.0\r\nContact:
\r\nTo:
;tag=47ed20c9\r\nFrom:
,
argv=0x7fff4bb823f8) at main.c:818
#17 main (argc=, argv=0x7fff4bb823f8) at main.c:1388
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Re: [OpenSIPS-Users] B2BUA Implementation for Transfer

[osiris123d]

OK.  So maybe the opensips core dumps were because I was calling the B2B_init
before I was calling my Nat_Traversal Route.  I placed my B2B_INIT after
doing NAT traversal and call transfers work almost ok.  I have a internal
user call a PSTN number, the call connects and then I transfer the PSTN call
to another internal user.  The call is transfered but the internal user that
gets the transfer gets a Caller ID of UNAVAILABLE.  The UNAVAILABLE is
coming from the PSTN SIP provider, so I am not sure if there is anything I
can do about that.

Here is my SIP trace of the call in action.  I also have a lot of NOTIFY
messages that appear to not get  any responses.  Am I doing this right
because non of my SIP Messages in this trace have a CALL-ID starting with
B2Bx.

U 2010/10/30 13:05:24.008082 75.65.8.158:2048 -> 173.203.87.134:5060
INVITE sip:19012138...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.33.22:2048;branch=z9hG4bK-7k2kuhfx9wy9;rport.
Route: .
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 3 INVITE.
Max-Forwards: 70.
Contact: ;reg-id=1.
X-Serialnumber: 0004132314FF.
P-Key-Flags: resolution="31x13", keys="4".
User-Agent: snom360/8.4.18.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Session-Expires: 3600;refresher=uas.
Min-SE: 90.
Content-Type: application/sdp.
Content-Length: 391.
.
v=0.
o=root 177494289 177494290 IN IP4 192.168.33.22.
s=call.
c=IN IP4 192.168.33.22.
t=0 0.
m=audio 56536 RTP/AVP 0 8 9 99 3 18 4 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:9 g722/8000.
a=rtpmap:99 g726-32/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 g729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 g723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendonly.

#
U 2010/10/30 13:05:24.008750 173.203.87.134:5060 -> 75.65.8.158:2048
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.33.22:2048;branch=z9hG4bK-7k2kuhfx9wy9;rport=2048;received=75.65.8.158.
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 3 INVITE.
Proxy-Authenticate: Digest realm="irock.com",
nonce="4ccc5e824843cab77333a72029c17bb4116ccb2d", qop="auth".
Server: OpenSIPS (1.6.3-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 2010/10/30 13:05:24.177906 75.65.8.158:2048 -> 173.203.87.134:5060
ACK sip:19012138...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.33.22:2048;branch=z9hG4bK-7k2kuhfx9wy9;rport.
Route: .
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 3 ACK.
Max-Forwards: 70.
Contact: ;reg-id=1.
Content-Length: 0.
.

#
U 2010/10/30 13:05:24.178269 173.203.87.134:5060 -> 64.2.142.15:5060
ACK sip:19012138...@64.2.142.15 SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 173.203.87.134;branch=z9hG4bK-7k2kuhfx9wy9.
Via: SIP/2.0/UDP
192.168.33.22:2048;received=75.65.8.158;branch=z9hG4bK-7k2kuhfx9wy9;rport=2048.
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 3 ACK.
Max-Forwards: 69.
Contact: ;reg-id=1.
Content-Length: 0.
.

#
U 2010/10/30 13:05:24.181713 75.65.8.158:2048 -> 173.203.87.134:5060
INVITE sip:19012138...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.33.22:2048;branch=z9hG4bK-6gmoef740jfr;rport.
Route: .
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 4 INVITE.
Max-Forwards: 70.
Contact: ;reg-id=1.
X-Serialnumber: 0004132314FF.
P-Key-Flags: resolution="31x13", keys="4".
User-Agent: snom360/8.4.18.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Session-Expires: 3600;refresher=uas.
Min-SE: 90.
Proxy-Authorization: Digest
username="9012732001",realm="irock.com",nonce="4ccc5e824843cab77333a72029c17bb4116ccb2d",uri="sip:19012138...@64.2.142.15",qop=auth,nc=0001,cnonce="459209a8",response="42a83400bbfd4447dda86e036f0fb4e6",algorithm=MD5.
Content-Type: application/sdp.
Content-Length: 391.
.
v=0.
o=root 177494289 177494290 IN IP4 192.168.33.22.
s=call.
c=IN IP4 192.168.33.22.
t=0 0.
m=audio 56536 RTP/AVP 0 8 9 99 3 18 4 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:9 g722/8000.
a=rtpmap:99 g726-32/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 g729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 g723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmt
#
U 2010/10/30 13:05:24.183488 173.203.87.134:5060 -> 75.65.8.158:2048
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP
192.168.33.22:2048;branch=z9hG4bK-6gmoef740jfr;rport=2048;received=75.65.8.158.
From: "Bob 2001" ;tag=hmcq5f4dwe.
To: ;tag=as1101def0.
Call-ID: 3c28d36e6dcb-zjx72y9kbluy.
CSeq: 4 INVITE.
Server: OpenSIPS (1.6.3-notls (x86_64/linux)).
Content-Length: 0.
.

#
U 2010/10/30 13:05:24.183685

Re: [OpenSIPS-Users] B2BUA Implementation for Transfer

[osiris123d]

Actually Anca for Transfers shouldn't I really be doing the following?

if(is_method("REFER") && !(src_ip == "OPENSIPS_IP") && !has_totag() ) {
{
location();
b2b_init_request("refer");
exit;
}


???

If I do it with "INVITE" instead of "REFER" OPENSIPS does a core dump
because of some Nat-Traversal stuff.
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[OpenSIPS-Users] B2BUA Implementation for Transfer

[osiris123d]

I have seen there are a couple of posts about this so I apologize if this
seems redundant.

I am starting to mess with the B2BUA and am having some issues.  I am trying
to solve the transfer issue when a local OpenSIPS subcriber is on the phone
with a PSTN user and wants to transfer the PSTN user to another OpenSIPS
user.  I have created the refer.xml file and set up the module parameters,
so all that is fine.  The issue I am seeing are the invites getting sent out
from OpenSIPS.

If I place my
b2b_init_request("refer");
at the very beginning of my route logic before consume_credentials(),
location() and all that other junk I see that two Invites are sent

Say the user is 4...@irock.com
The B2B CALL-ID generated message gets sent to the actual IP address of
irock.com, which it shouldn't.
Then a correct invite gets sent to 4...@7x.45.x.44 which is the correct info
that is pulled from the location table, but this invite does not have the
B2B CALL-ID info in it.  So it seems I have placed the b2b_init_request in
the wrong spot on my script.


Then when I place the b2b_init_request after I execute the location() in my
route logic I see the following
two invites

The first invite is the B2B invite with the B2B CALL-ID.  It gets sent
correctly to 4...@7x.45.x.44 instead of to the actually IP of irock.com.  The
only problem here is that within the actual TO: field of the sip message it
is 4...@7x.45.x.44 instead of 4...@irock.com.  So the phone rejects this
invite with a "SIP/2.0 404 Not Found" since the phone only knows itself as
4...@irock.com.

The second invite is as usual the normal invite without the B2B CALL-ID.

So I guess the issue I am having is that dual invites are being sent and the
B2B is either sending it to the wrong IP address or it is placing the wrong
domain in the invite for the user.
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Re: [OpenSIPS-Users] REFER - Transferring Best Practice.

[osiris123d]

So excuse me for my ignorance but are we saying that OpenSIPS B2BUA can allow
an OpenSIPS subcriber to be on a call with a PSTN user and transfer to
someone else if configured correctly without needing a Asterisk box?


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Re: [OpenSIPS-Users] OpenSIPS hang - 1.6.3

[osiris123d]

A long time ago I was working with PUA and Presence and I started having
issues.  I upgraded to the latest version when I heard about the deadlock,
but I am still noticing from Monit that stuff freezes up.  I usually have to
restart the opensips process and all is better again.  I am reinstalling a
fresh version of OpenSIPS on a new VM, but I swear when I started messing
with PUA to get Call Pickup and BLF my box started having issues.

We shall see with the new box.  Hopefully it was something else.
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Re: [OpenSIPS-Users] Hunt Group with Opensips

[osiris123d]

Whoops I think I meant "longest Idle".  This is how cisco defines the option

Longest Idle—Calls go to the directory number that has
been idle for the longest time, according to the time
stamp of the most recent call to the hunt group taken by
that extension. If that extension is unavailable, the search
continues to the next extension in the group.


That really shouldn't be that hard.  All you would have to do is save all
the timestamps of the last call for each member, then just search for the
one with the oldest time.  Sweet.
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Re: [OpenSIPS-Users] Hunt Group with Opensips

[osiris123d]

Whoops I meant Brad.  Thanks.  I will try to figure out how to do least
called->gets called and post.
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Re: [OpenSIPS-Users] Hunt Group with Opensips

[osiris123d]

Deon,

Just wondering why you set the next hunt number twice in your else
condition??


  } else {
#Not the 1st time, let's pull the next in order
$avp(s:hunt_num) = $(avp(s:hunt_list){s.select,$avp(s:hunt_pos),,});
xlog("L_NOTICE","***Trying #$avp(s:hunt_pos) of $avp(s:total_hunts) in
seq*** Number to hunt to: $avp(s:hunt_num) Call ID: $ci with Source ANI of:
$fU");
#pull the next number to call from the set 
$avp(s:hunt_num) = $(avp(s:hunt_list){s.select,$avp(s:hunt_pos),,});

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[OpenSIPS-Users] Hunt Group with Opensips

[osiris123d]

I saw a post saying it was possible to do hunt groups with col, but i wasn't
sure how one would keep up with calls to each group user so that one could
do "least called user" or even the user that was next in line to be called. 
I am guessing avps and contact array could help, but i was just wondering
what solutions some people might have used.  I was even thinking that i
might be able to relay the calls to a hunt group number to asterisk and let
asterisk handle the hunt group logic, but i am not sure if that would work.
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[OpenSIPS-Users] Opensips 1.6.3 Crashing: db_mysql_val2bind bug?

[osiris123d]

I just upgraded to 1.6.3 to fix a bug I was noticing with pua_dialoginfo. 
1.6.3 appears to have fixed that issue but I am noticing that it is now
randomly crashing.  Here is a core dump

Proxy01:/usr/local/sbin# gdb opensips core
GNU gdb 6.8-debian
Copyright (C) 2008 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later

This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu"...

warning: Can't read pathname for load map: Input/output error.
Reading symbols from /lib/libdl.so.2...done.
Loaded symbols for /lib/libdl.so.2
Reading symbols from /lib/libresolv.so.2...done.
Loaded symbols for /lib/libresolv.so.2
Reading symbols from /lib/libc.so.6...done.
Loaded symbols for /lib/libc.so.6
Reading symbols from /lib/ld-linux-x86-64.so.2...done.
Loaded symbols for /lib64/ld-linux-x86-64.so.2
Reading symbols from /usr/local/lib64/opensips/modules/db_mysql.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/db_mysql.so
Reading symbols from /usr/lib/libmysqlclient.so.15...done.
Loaded symbols for /usr/lib/libmysqlclient.so.15
Reading symbols from /lib/libpthread.so.0...done.
Loaded symbols for /lib/libpthread.so.0
Reading symbols from /lib/libcrypt.so.1...done.
Loaded symbols for /lib/libcrypt.so.1
Reading symbols from /lib/libnsl.so.1...done.
Loaded symbols for /lib/libnsl.so.1
Reading symbols from /lib/libm.so.6...done.
Loaded symbols for /lib/libm.so.6
Reading symbols from /usr/lib/libz.so.1...done.
Loaded symbols for /usr/lib/libz.so.1
Reading symbols from /usr/local/lib64/opensips/modules/signaling.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/signaling.so
Reading symbols from /usr/local/lib64/opensips/modules/sl.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/sl.so
Reading symbols from /usr/local/lib64/opensips/modules/tm.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/tm.so
Reading symbols from /usr/local/lib64/opensips/modules/rr.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/rr.so
Reading symbols from /usr/local/lib64/opensips/modules/uac.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/uac.so
Reading symbols from /usr/local/lib64/opensips/modules/maxfwd.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/maxfwd.so
Reading symbols from /usr/local/lib64/opensips/modules/usrloc.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/usrloc.so
Reading symbols from /usr/local/lib64/opensips/modules/registrar.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/registrar.so
Reading symbols from /usr/local/lib64/opensips/modules/textops.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/textops.so
Reading symbols from /usr/local/lib64/opensips/modules/mi_fifo.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/mi_fifo.so
Reading symbols from
/usr/local/lib64/opensips/modules/mi_datagram.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/mi_datagram.so
Reading symbols from /usr/local/lib64/opensips/modules/uri.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/uri.so
Reading symbols from /usr/local/lib64/opensips/modules/acc.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/acc.so
Reading symbols from /usr/local/lib64/opensips/modules/avpops.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/avpops.so
Reading symbols from /usr/local/lib64/opensips/modules/auth.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/auth.so
Reading symbols from /usr/local/lib64/opensips/modules/auth_db.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/auth_db.so
Reading symbols from /usr/local/lib64/opensips/modules/alias_db.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/alias_db.so
Reading symbols from /usr/local/lib64/opensips/modules/domain.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/domain.so
Reading symbols from /usr/local/lib64/opensips/modules/group.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/group.so
Reading symbols from /usr/local/lib64/opensips/modules/presence.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/presence.so
Reading symbols from /usr/lib/libxml2.so.2...done.
Loaded symbols for /usr/lib/libxml2.so.2
Reading symbols from
/usr/local/lib64/opensips/modules/presence_mwi.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/presence_mwi.so
Reading symbols from
/usr/local/lib64/opensips/modules/presence_xml.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/presence_xml.so
Reading symbols from
/usr/local/lib64/opensips/modules/presence_dialoginfo.so...done.
Loaded symbols for /usr/local/lib64/opensips/modules/presence_dialoginfo.so
Reading symbols from /usr/local/lib64/opensips/modules/p

[OpenSIPS-Users] pua_dialoginfo SVN 6958 causing crashes

[osiris123d]

I recently compiled pua_dialoginfo SVN 6958 so that I could use the caller
and callee _spec_param.  Seems that ever since I started using that my
OpenSIPS process has been crashing a lot.  Wanted to capture a syslog since
1.6.3 is coming out soon.  I finally caught a debug=6 syslog and it shows
the following

Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:tm:run_trans_callbacks: trans=0x7f0fdd761da0, callback type 128, id 0 e
ntered
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:dialog:next_state_dlg: dialog 0x7f0fdd75c610 changed from state 1 to st
ate 2, due event 2
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:dialog:run_dlg_callbacks: dialog=0x7f0fdd75c610, type=128
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:dialog:run_dlg_callbacks: dialog=0x7f0fdd75c610, type=128
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:dialog:fetch_dlg_value: looking for 
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:dialog:fetch_dlg_value: var found-> !
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:pua_dialoginfo:__dialog_sendpublish: peer_uri = sip:8005558...@irock.co
m;user=phone#015#012
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27286]:
DBG:core:parse_to_param: user=phone
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27300]:
CRITICAL:core:receive_fd: EOF on 6
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27300]:
DBG:core:handle_ser_child: dead child 1, pid 27286 (shutting down?)
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27300]:
DBG:core:io_watch_del: io_watch_del (0x74e920, 6, -1, 0x0) fd_no=18 called
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27285]:
INFO:core:handle_sigs: child process 27286 exited by a signal 11
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27285]:
INFO:core:handle_sigs: core was not generated
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27285]:
INFO:core:handle_sigs: terminating due to SIGCHLD
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27298]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 kernel: [445173.921384] opensips[27286]: segfault at
28 ip 004cac06 sp 7fff4401ddb0 error 6 in o
pensips[40+14]
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27300]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27297]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27295]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27289]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27294]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27296]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27299]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27291]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27288]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27290]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27292]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27287]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27293]: INFO:core:sig_usr:
signal 15 received
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was closed cleanly.
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27285]: DBG:rls:destroy:
start
Jul 11 19:22:08 Proxy01 /usr/local/sbin/opensips[27285]:
DBG:presence:update_db_subs: delete expired
Jul 11 19:22:08 Proxy01 media-dispatcher[2778]: debug: Connection to
OpenSIPS lost: Connection was c

Re: [OpenSIPS-Users] Call Pickup not working for Outbound to Inbound calls

[osiris123d]

You are absolutely right.

Here is the sip message I posted yesterday of User B trying to grab the call

U 2010/07/05 21:35:23.270298 173.x.x.134:5060 -> 64.2.142.93:5060 
INVITE sip:[hidden email]:5060 SIP/2.0. 
Record-Route: 
. 
Via: SIP/2.0/UDP 173.x.x.134;branch=z9hG4bK104b.eef8e637.0. 
Via: SIP/2.0/UDP 
192.168.0.12:2077;received=75.x.x.158;branch=z9hG4bK-whzk6bav7ueb;rport=1025. 
From: "Blah 2001" ;tag=oydxjqj42f. 
To: . 
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5. 
CSeq: 2 INVITE. 
Max-Forwards: 69. 
Contact: ;reg-id=1. 
Replaces: 
[hidden email];to-tag=as55840b13;from-tag=5667c613. 


And here is the actual first invite that comes from the PSTN that I didn't
originally post

U 2010/07/05 21:35:18.642951 64.2.142.15:5060 -> 173.x.x.134:5060
INVITE sip:9xx2xx2x...@173.x.x.134 SIP/2.0.
Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK19694dc7;rport.
From: "9xx8xx3xx2" ;tag=as55840b13.
To: .
Contact: .
Call-ID: 4df4f89829fad0997025b2d5532d7...@64.2.142.15.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Tue, 06 Jul 2010 02:35:19 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 332.
.
v=0.
o=root 15670 15670 IN IP4 64.2.142.15.
s=session.
c=IN IP4 64.2.142.15.
t=0 0.
m=audio 14536 RTP/AVP 0 8 3 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.



So the first initial invite comes from Vitelity's 64.2.142.15 Asterisk PBX,
but my Call Pickup Invite gets sent to Vitelity's 64.2.142.93 Asterisk PBX. 
I will have to play with this in the script logic to make sure the invite is
sent back to the original requestor.

Thanks for the clue.
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[OpenSIPS-Users] Call Pickup not working for Outbound to Inbound calls

[osiris123d]

I successfully have Call Pickup working between internal users but when I
have a PSTN user call Internal User A and then Internal User B tries to do a
call pickup I get a "481 call leg does not exist".  When the PSTN call comes
in from my Vitelity SIP provider I see that they support the following

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.

Is what I am trying to do possible?  Here is a sip trace of the issue (it
starts with Internal User B sending the invite to try and pickup the call
from the PSTN user to User A)

U 2010/07/05 21:35:23.016955 75.x.x.158:1025 -> 173.x.x.134:5060
INVITE sip:9xx83x3...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.12:2077;branch=z9hG4bK-2peqfmxxyu5c;rport.
From: "Blah 2001" ;tag=oydxjqj42f.
To: .
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5.
CSeq: 1 INVITE.
Max-Forwards: 70.
Contact: ;reg-id=1.
Replaces:
4df4f89829fad0997025b2d5532d7...@64.2.142.15;to-tag=as55840b13;from-tag=5667c613.
X-Serialnumber: 0004132314FF.
P-Key-Flags: resolution="31x13", keys="4".
User-Agent: snom360/8.2.29.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Session-Expires: 3600;refresher=uas.
Min-SE: 90.
Content-Type: application/sdp.
Content-Length: 391.
.
v=0.
o=root 454576176 454576176 IN IP4 192.168.0.12.
s=call.
c=IN IP4 192.168.0.12.
t=0 0.
m=audio 50978 RTP/AVP 0 8 9 103 3 18 4 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:9 g722/8000.
a=rtpmap:103 g726-32/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 g729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 g723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 2010/07/05 21:35:23.018808 173.x.x.134:5060 -> 75.x.x.158:1025
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.0.12:2077;branch=z9hG4bK-2peqfmxxyu5c;rport=1025;received=75.x.x.158.
From: "Blah 2001" ;tag=oydxjqj42f.
To: ;tag=fa9ad7a35cf4c9d5a26f33f3220f505e.db11.
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5.
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="irock.com",
nonce="4c32968970613f35e44f447288865f39b1af9c25", qop="auth".
Server: AE SIP Proxy.
Content-Length: 0.
.

#
U 2010/07/05 21:35:23.092585 75.x.x.158:1025 -> 173.x.x.134:5060
ACK sip:9xx83x3...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.12:2077;branch=z9hG4bK-2peqfmxxyu5c;rport.
From: "Blah 2001" ;tag=oydxjqj42f.
To: ;tag=fa9ad7a35cf4c9d5a26f33f3220f505e.db11.
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5.
CSeq: 1 ACK.
Max-Forwards: 70.
Contact: ;reg-id=1.
Content-Length: 0.
.

#
U 2010/07/05 21:35:23.109251 75.x.x.158:1025 -> 173.x.x.134:5060
INVITE sip:9xx83x3...@64.2.142.15 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.12:2077;branch=z9hG4bK-whzk6bav7ueb;rport.
From: "Blah 2001" ;tag=oydxjqj42f.
To: .
Call-ID: 3c27c0c40a8e-c8exk4s3gwo5.
CSeq: 2 INVITE.
Max-Forwards: 70.
Contact: ;reg-id=1.
Replaces:
4df4f89829fad0997025b2d5532d7...@64.2.142.15;to-tag=as55840b13;from-tag=5667c613.
X-Serialnumber: 0004132314FF.
P-Key-Flags: resolution="31x13", keys="4".
User-Agent: snom360/8.2.29.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Session-Expires: 3600;refresher=uas.
Min-SE: 90.
Proxy-Authorization: Digest
username="9xx27x2xx1",realm="irock.com",nonce="4c32968970613f35e44f447288865f39b1af9c25",uri="sip:9xx83x3...@64.2.142.15",qop=auth,nc=0001,cnonce="79abe95f",response="1fd49a7fdf2da6c8f8e5728bc69b49fc",algorithm=MD5.
Content-Type: application/sdp.
Content-Length: 391.
.
v=0.
o=root 454576176 454576176 IN IP4 192.168.0.12.
s=call.
c=IN IP4 192.168.0.12.
t=0 0.
m=audio 50978 RTP/AVP 0 8 9 103 3 18 4 101.
a=rtpmap:0 pcmu/8000.
a=rtpmap:8 pcma/8000.
a=rtpmap:9 g722/8000.
a=rtpmap:103 g726-32/8000.
a=rtpmap:3 gsm/8000.
a=rtpmap:18 g729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:4 g723/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
#
U 2010/07/05 21:35:23.129530 173.x.x.134:5060 -> 173.x.x.134:5060
PUBLISH sip:9xx27x2...@irock.com SIP/2.0.
Via: SIP/2.0/UDP 173.x.x.134;branch=z9hG4bK3d0b.b751b896.0.
To: sip:9xx27x2...@irock.com.
From: ;tag=144b207508c0e0296792f226e399eaf5-1cb8.
CSeq: 10 PUBLISH.
Call-ID: 41d706004e80f741-32...@173.x.x.134.
Content-Length: 485.
User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)).
Max-Forwards: 70.
Event: dialog.
Expires: 3601.
Content-Type: application/dialog-info+xml.
.

tryingsip:9xx83x3...@64.2.142.15sip:9xx27x2...@irock.com

#
U 2010/07/05 21:35:23.134007 173.x.x.134:5060 -> 173.x.x.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 173.x.x.134;branch=z9hG4bK3d0b.b751b896.0.
To: sip:9xx27x2...@irock.com;tag=a93952c7ce25f579a59b9db9d9acf225-6f03.
From: ;tag=144b207508c0e0296792f226e399eaf5-1cb8.
CSeq: 10 PUBLISH.
Call-ID: 41d706004e80f741-32...@173.x.x.134.
Expires: 3600.
SIP-ETag: a.1278333080.32345.74.0.
S

Re: [OpenSIPS-Users] PUA Dialoginfo - BLF with Multiple Domain

[osiris123d]

Thanks for the info.

On Jun 29, 2010 4:37am, "Anca Vamanu-2 [via OpenSIPS (Open SIP Server)]"  
 wrote:


















> On 06/29/2010 04:56 AM, osiris123d wrote:





> So I have it working now but noticed one issue.



> I allow my users to dial without entering the areacode. I prepend the area

> code thanks to avps. The issue I am seeing is that BLF doesn't work on  
> some

> calls. I know this is due to the fact that the dialog appears different

> because the TO header is missing the prepended area code. From the PUA

> Dialoginfo documentation page I see that there is a callee_spec_param

> parameter that I can set. When I add this to my script and try to start up

> OpenSIPS I am getting the following error



> Not starting opensips: invalid configuration file!



> Jun 28 20:48:55 [15871] ERROR:core:set_mod_param_regex: parameter

> not found in module

> Jun 28 20:48:55 [15871] CRITICAL:core:yyerror: parse error in config file,

> line 286, column 20-21: Parameter not found in module

> - can't set

> Jun 28 20:48:55 [15871] ERROR:core:main: bad config file (1 errors)







> This is what I have on line 286 of my config

> modparam("pua_dialoginfo", "callee_spec_param", "$avp(i:20)")



> I am running opensips-1.6.2-tls



> What could I be doing wrong?






> Hi,


> This feature is not available in 1.6.2 but only in

> svn version. I advise you to take the pua_dialoginfo module from there.





> Regards,




> --

> Anca Vamanu

> www.voice-system.ro








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> To unsubscribe from Re: PUA Dialoginfo - BLF with Multiple Domain, click  
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Re: [OpenSIPS-Users] Opensips+Multiple backup servers

[osiris123d]

I believe it will also depend on whether or not your phone clients are behind
a firewall or not.  If they are then you will have to worry about the fact
that if the OpenSIPS proxy that the client first registered with goes down
then the backup OpenSIPS proxy will not be able to send the client messages
since the NAT Firewall doesn't have a pinhole opened up for that backup
OpenSIPS router (this assumes that the backup OpenSIPS router has a
different IP address from the primary).  A way around this would be HA. 
Another option would be if the NAT Firewall/WAN Router has some kind of SIP
functions like the Edgewater Network EdgeMarc devices.

Another option is the fact that most IP phones can be configured with a
failover Registration Server IP.
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Re: [OpenSIPS-Users] PUA Dialoginfo - BLF with Multiple Domain

[osiris123d]

So I have it working now but noticed one issue.

I allow my users to dial without entering the areacode.  I prepend the area
code thanks to avps.  The issue I am seeing is that BLF doesn't work on some
calls.  I know this is due to the fact that the dialog appears different
because the TO header is missing the prepended area code.  From the PUA
Dialoginfo documentation page I see that there is a callee_spec_param
parameter that I can set.  When I add this to my script and try to start up
OpenSIPS I am getting the following error

Not starting opensips: invalid configuration file!

Jun 28 20:48:55 [15871] ERROR:core:set_mod_param_regex: parameter
 not found in module 
Jun 28 20:48:55 [15871] CRITICAL:core:yyerror: parse error in config file,
line 286, column 20-21: Parameter  not found in module
 - can't set
Jun 28 20:48:55 [15871] ERROR:core:main: bad config file (1 errors)



This is what I have on line 286 of my config
modparam("pua_dialoginfo", "callee_spec_param", "$avp(i:20)")

I am running opensips-1.6.2-tls

What could I be doing wrong?
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Re: [OpenSIPS-Users] PUA Dialoginfo - BLF with Multiple Domain

[osiris123d]

Excellent.  I will edit my script and test.

Thanks.

On Tue, Jun 15, 2010 at 3:08 AM, Anca Vamanu-2 [via OpenSIPS (Open SIP
Server)] 

> wrote:

> Hi,
>
> In fact this limitation is no longer valid. I have changed this a couple
> of months ago - when the possiblity to choose for which party of the
> dialog to publish state was added. I updated the documentation then but
> did not notice this entry in the limitations list. Good for you that you
> posted this question. I will update the documentation
> So now the power is in the hands of the script writer(as it is normal) -
> he has to decide whether the caller and callee are local and call
> dialoinfo_set for one or both.
>
> http://www.opensips.org/html/docs/modules/devel/pua_dialoginfo.html#id228400
>
> Regards,
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
> On 06/15/2010 04:02 AM, osiris123d wrote:
> > I was wondering if anything was in the works for multiple domain support
> with
> > PUA Dialoginfo?
> >
>
>
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[OpenSIPS-Users] PUA Dialoginfo - BLF with Multiple Domain

[osiris123d]

I was wondering if anything was in the works for multiple domain support with
PUA Dialoginfo?

On the documentation page for PUA Dialoginfo it says the following

The module tries to find out if the entity is a local user. Only PUBLISH to
local user are sent. Therefore, the module needs to find out if the domain
is a local one or not. It uses the same mechanism as the "==myself"
mechanism. Thus, all domains have to be declared with the "alias=..." option
in OpenSIPS.cfg. DB-based multidomain support as offered by "domain" module
is not supported yet. Conclusion: The module has the same "domain" problems
as the "rr" module. 
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

Forgot about that. Thanks for the info.



On Jun 3, 2010 2:03pm, "Adrian Georgescu [via OpenSIPS (Open SIP Server)]"  
 wrote:




> You can use soap-simple-proxy to build a web page where pres-rules  
> permissions are managed.



> Adrian




> On Jun 3, 2010, at 4:27 PM, osiris123d wrote:



> >


> > If I can ever get Snom working with OpenXCAP the next thing I am going  
> to


> > wonder is how permissions are handled. I know with Counterpath Bria when


> > you add someone as a buddy the buddy gets a message asking if he/she  
> would


> > like to allow this person to view their presence.


> >


> > I am seriously thinking that Snom will not have something like this and  
> I


> > will be forced to set force_active=1 (not what I prefer, but at least  
> its a


> > start).


> > --


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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

If I can ever get Snom working with OpenXCAP the next thing I am going to
wonder is how permissions are handled.  I know with Counterpath Bria when
you add someone as a buddy the buddy gets a message asking if he/she would
like to allow this person to view their presence.

I am seriously thinking that Snom will not have something like this and I
will be forced to set force_active=1 (not what I prefer, but at least its a
start).
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

Sweet.  I might have found my first bug on this forum .

I went ahead and posted my findings on Snom's forum today.  I will see what
they say.  Also would like to hear what you find after checking.

Thanks for keeping up with this.
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

Sorry Saul,

I got my Sniff's messed up.  It should have been the other way around
(Should have double checked my message).

Attached is a txt file with both the Firefox and Snom sniff.  I have also
included a good Counterpath Bria sniff from a different user.  From the
looks of it the difference here for the good HTTP GETS is that the GET
starts with the /xcap-r...@domain.com/ path for the URI, whereas the SNOM
phone starts with
https://xcap.server.com:443/xcap-r...@domain.com/<https://xcap.server.com/xcap-r...@domain.com/>path.

I don't think the Snom phone has an HTTP trace tool.  The snom can do sniffs
and save it in .pcap format.  I can try and see what kind of output I get
from that.

On Wed, Jun 2, 2010 at 2:02 AM, Saúl Ibarra Corretgé [via OpenSIPS (Open SIP
Server)] 

> wrote:

> Hi,
>
> On 02/06/10 05:01, osiris123d wrote:
> >
> > I still can't do a ngrep capture since ngrep can't decode SSL as far as I
>
> > know, but I can type out a bad HTTP GET request from SNOM and also a good
>
> > HTTP GET request from Firefox.  I think you will notice where the
> difference
> > is and I am not sure if OpenXCAP is getting confused by the difference or
> if
> > SNOM's code is wrong when it comes to XCAP GETS.  There has to be an
> answer
> > in here somewhere.
> >
>
> I'm a bit confused about your email, see inline comments.
>
> >
> > GOOD Firefox GET request (rewritten from Wireshark capture)
> > 
> > GET
> >
> https://xcap.ae.com:443/xcap-r...@.../resource-lists/users/sip:9xx2xx2...@.../index<https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index>
>
> > HTTP/1.1\r\n
> > Expert Info (Chat/Sequence): GET
> >
> https://xcap.ae.com:443/xcap-r...@.../resource-lists/users/sip:9xx2xx2...@.../index<https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index>
>
> > HTTP/1.1\r\n
> > Message: GET
> >
> https://xcap.ae.com:443/xcap-r...@.../resource-lists/users/sip:9xx2xx2...@.../index<https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index>
>
> > HTTP/1.1\r\n
> > Severity level: Chat
> > Group: Sequence
> > Request Method: GET
> > Request URI:
> >
> https://xcap.ae.com:443/xcap-r...@.../resource-lists/users/sip:9xx2xx2...@.../index<https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index>
>
> > Request Version: HTTP/1.1
> > Host: xcap.ae.com:443\r\n
> > Content-Length: 0\r\n
> > Content length: 0
> > Accept-Language: en\r\n
> > Connection: Keep-Alive\r\n
> > Keep-Alive: 5\r\n
> > User-Agent: Mozilla/4.0 (compatible; snom360-SIP 8.2.29 1.1.3-m)\r\n
> > \r\n
>
> If these are the good requests why is there a Snom useragent?
>
> >
> >
> >
> >
> > BAD SNOM GET request (rewritten from Wireshark capture)
> > 
> > GET /[hidden email]<http://user/SendEmail.jtp?type=node&node=5129554&i=0>
> /resource-lists/users/sip:[hidden 
> email]<http://user/SendEmail.jtp?type=node&node=5129554&i=1>/index
>
> > HTTP/1.1\r\n
> > Expert Info (Chat/Sequence): GET
> > /[hidden email] <http://user/SendEmail.jtp?type=node&node=5129554&i=2>
> /resource-lists/users/sip:[hidden 
> email]<http://user/SendEmail.jtp?type=node&node=5129554&i=3>/index
>
> > HTTP/1.1\r\n
> > Message: GET
> > /[hidden email] <http://user/SendEmail.jtp?type=node&node=5129554&i=4>
> /resource-lists/users/sip:[hidden 
> email]<http://user/SendEmail.jtp?type=node&node=5129554&i=5>/index
>
> > HTTP/1.1\r\n
> > Severity level: Chat
> > Group: Sequence
> > Request Method: GET
> > Request URI:
> > /[hidden email] <http://user/SendEmail.jtp?type=node&node=5129554&i=6>
> /resource-lists/users/sip:[hidden 
> email]<http://user/SendEmail.jtp?type=node&node=5129554&i=7>/index
>
> > Request Version: HTTP/1.1
> > Host: xcap.ae.com\r\n
> > User-Agent: Mozilla/5.0 (Windows; U; Windows NT 6.1; en-US; rv:1.9.2.3)
> > Gecko/20100401 Firefox/3.6.3\r\n
> > Accept:
> text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8\r\n
> > Accept-Language: en-us,en;q=0.5\r\n
> > Accept-Encoding: gzip,deflate\r\n
> > Accept-Charset: ISO-8859-1,utf-8;q=0.7,*;q=0.7\r\n
> > Keep-Alive: 115\r\n
> > Connection: keep-alive\r\n
> > Cookie: __utma=87383048.501874217.1266983378.1266983378.1266983378.1;
> >
> __utmz=873

Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

I still can't do a ngrep capture since ngrep can't decode SSL as far as I
know, but I can type out a bad HTTP GET request from SNOM and also a good
HTTP GET request from Firefox.  I think you will notice where the difference
is and I am not sure if OpenXCAP is getting confused by the difference or if
SNOM's code is wrong when it comes to XCAP GETS.  There has to be an answer
in here somewhere.


GOOD Firefox GET request (rewritten from Wireshark capture)

GET
https://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Expert Info (Chat/Sequence): GET
https://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Message: GET
https://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Severity level: Chat
Group: Sequence
Request Method: GET
Request URI:
https://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
Request Version: HTTP/1.1
Host: xcap.ae.com:443\r\n
Content-Length: 0\r\n
Content length: 0
Accept-Language: en\r\n
Connection: Keep-Alive\r\n
Keep-Alive: 5\r\n
User-Agent: Mozilla/4.0 (compatible; snom360-SIP 8.2.29 1.1.3-m)\r\n
\r\n




BAD SNOM GET request (rewritten from Wireshark capture)

GET /xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Expert Info (Chat/Sequence): GET
/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Message: GET
/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1\r\n
Severity level: Chat
Group: Sequence
Request Method: GET
Request URI:
/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
Request Version: HTTP/1.1
Host: xcap.ae.com\r\n
User-Agent: Mozilla/5.0 (Windows; U; Windows NT 6.1; en-US; rv:1.9.2.3)
Gecko/20100401 Firefox/3.6.3\r\n
Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8\r\n
Accept-Language: en-us,en;q=0.5\r\n
Accept-Encoding: gzip,deflate\r\n
Accept-Charset: ISO-8859-1,utf-8;q=0.7,*;q=0.7\r\n
Keep-Alive: 115\r\n
Connection: keep-alive\r\n
Cookie: __utma=87383048.501874217.1266983378.1266983378.1266983378.1;
__utmz=87383048.1266983378.1.1.utmccn=(direct)|utmcsr=(direct)|utmcmd=(none)\r\n
\r\n
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

Saul/Adrian,

I did paste the URL into a browser and everything worked out fine. When I  
get home I will do more testing with the Snom phone because I have no clue  
why I am seeing that double url.

Also here is my OpenXCAP config

;
; Configuration file for OpenXCAP
;
; Copyright (c) 2007-2009 AG Projects
; http://ag-projects.com

[Server]

; IP address and port to listen for requests
; 0.0.0.0 means any address of this host
address = 17X.XXX.XXX.XXX
port = 443
;port = 80
; The XCAP Root URI; must not contain any port number. If the scheme is
; https, then the server will listen for requests in TLS mode.

; This is a comma/space separated list of XCAP root URIs. The first is the
; primary XCAP root URI, while the others (if specified) are aliases.
; The primary root URI is used when generating xcap-diff

root = https://xcap.ae.com/xcap-root
;root = http://xcap.ae.com/xcap-root
; The backend to be used for storage and authentication. Current supported
; values are Database and OpenSIPS. OpenSIPS backend inherits all the  
settings
; from the Database backend but performs extra actions related to the
; integration with OpenSIPS for which it read the settings from [OpenSIPS]
; section

backend = OpenSIPS

; Validate XCAP documents against XML schemas
document_validation = Yes


[Logging]
; directory where to put log files, default is /var/log/openxcap
; if empty, like in the following line, logs will be printed to stdout
directory= /var/log/openxcap

; The following log_* parameters control what information is logged for
; which errors. Their format is comma-separated list of HTTP error codes
; that should enable the feature currently implemented: log_request_headers,
; log_response_body, log_stacktrace

; Some examples:

; * log stack trace for 500 Internal Error only (default)
log_stacktrace=500

; * log stack trace (if available) for any error
log_stacktrace=any

; * log responses sent to the client for 400 and 409 errors (default is 500)
log_response_body=400,409

; * log headers sent by the client for 401 errors (default is 500)
log_request_headers=401


[Authentication]

; The HTTP authentication type, this can be either 'basic' or 'digest'.
; If you're using TLS, it's better to choose 'basic' because the data is
; encrypted anyway.
type = basic

; Specify if the passwords are stored as plain text - Yes
; or in a hashed format MD5('username:domain:password') - No
cleartext_passwords = No

; The default authentication realm
default_realm = ae.com

; A list of trusted peers from where XCAP requests are accepted without HTTP
; authentication eg trusted_peers = 10.0.0.0/24, 192.168.0.1
trusted_peers = 66.XX.XXX.XXX, 127.0.0.1, 17X.XXX.XXX.XXX


[TLS]

; Location of X509 certificate and private key that identify this server.
; The path is relative to /etc/openxcap, or it can be given as an absolute
; path.

; Server X509 certificate
certificate = tls/openxcapserver.crt

; Server X509 private key
private_key = tls/openxcapserver.key


[Database]

; The database connection URI for the datase with subscriber accounts
authentication_db_uri = mysql://opensips:...@17x.xxx.xxx.xxx/opensips

; The database connection URI for the database that stores the XCAP  
documents
storage_db_uri = mysql://opensips:xx...@17x.xxx.xxx.xxx/opensips

; Authentication and storage tables
subscriber_table = subscriber
xcap_table = xcap


[OpenSIPS]

; The address and port of the xml-rpc management interface
xmlrpc_url = http://17X.XXX.XXX.XXX:8080

; Publish xcap-diff event via OpenSIPS management interface
enable_publish_xcapdiff = yes

On May 27, 2010 1:35am, "Saúl Ibarra Corretgé [via OpenSIPS (Open SIP  
Server)]"  wrote:




> Hi,



> On 27/5/10 4:21 AM, osiris123d wrote:


> >


> > I set up Snom to work off of port 80 and the ngrep capture of port 80  
> on the


> > openxcap server is


> >


> > 


> > T 2010/05/26 20:28:39.863832 7X.XX.8.XX8:2108 -> 17X.XXX.XXX.XX4:80 [AP]


> > ?...;(m..Jdb...


> > 


> >


> > And the snom log says


> > [1] 23/12/2001 18:01:08: webclient: request


> > xcap.ae.com:80https://xcap.ae.com:[hidden  
> email]/resource-lists/users/sip:[hidden email]/index


> > stopped due to no response from server


> > [5] 23/12/2001 18:01:08: XCAPclient: server did not provide an etag


> > [5] 23/12/2001 18:01:08: XCAPclient: Retrieving directory failed with


> > error-code 408 and message:


> >


> >


> > I am not able to get a clear ngrep capture but I can tell you for a fact


> > that on the client side I am not sending a doubled up url of


> > https://xcap.ae.comhttps://xcap.ae.com
> >



> Looks like the Snom uses SSL anyway.



> > I believe the first line in the openxcap access.log verifies this.


> >


> > 7X.XX.

Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

I set up Snom to work off of port 80 and the ngrep capture of port 80 on the
openxcap server is


T 2010/05/26 20:28:39.863832 7X.XX.8.XX8:2108 -> 17X.XXX.XXX.XX4:80 [AP]
?...;..<&p&!\...\...+...(m..Jd.b


And the snom log says
[1] 23/12/2001 18:01:08: webclient: request
xcap.ae.com:80https://xcap.ae.com:80/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
stopped due to no response from server
[5] 23/12/2001 18:01:08: XCAPclient: server did not provide an etag
[5] 23/12/2001 18:01:08: XCAPclient: Retrieving directory failed with
error-code 408 and message:


I am not able to get a clear ngrep capture but I can tell you for a fact
that on the client side I am not sending a doubled up url of
https://xcap.ae.comhttps://xcap.ae.com

I believe the first line in the openxcap access.log verifies this.

7X.XX.XX.XX8 'GET
https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx27x2...@irock.com/index
HTTP/
1.1' 404 0 179 'Mozilla/4.0 (compatible; snom360-SIP 8.2.29 1.1.3-m)' -
TRACEBACK (most recent call last):
  File
"/usr/lib/python2.5/site-packages/twisted/web2/channel/http.py", line 412,
in processRequest
self.request.process()
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 299, in process
d.callback(None)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 280, in callback
self._startRunCallbacks(result)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 354, in _startRunCallbacks
self._runCallbacks()
---  ---
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 371, in _runCallbacks
self.result = callback(self.result, *args, **kw)
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 296, in 
d.addCallback(lambda res, req: res.renderHTTP(req), self)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
335, in renderHTTP
d = self.authenticate(request)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
225, in authenticate
xcap_uri = parseNodeURI(uri, AuthenticationConfig.default_realm)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
94, in parseNodeURI
raise ResourceNotFound("XCAP root not found for URI: %s" %
node_uri)
xcap.errors.ResourceNotFound: XCAP root not found for URI:
https://xcap.ae.comhttps://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx27x2...@irock.com/index


I am running the OpenXCAP 1.2.0


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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

I will try and do that later today.  I think i tried to disable HTTPS when I
first started testing with the Snom phone, but I believe the Snom phone
always tries to use HTTPS.  If you notice in the URL path that the Snom
sends to OpenXCAP it automatically assumes https://servername then it will
attach the port that you provide in the Snom xcap port setting.  So then you
see https://servername:443 <https://servername/>.  I am worried that if I
set the Snom XCAP port to be 80 then the url will be https://servername:80

We shall see.

Adrian Georgescu,

I do realize that the URI that OpenXCAP is seeing is repeated
https://xcap.ae.comhttps://xcap.ae.com:443<https://xcap.ae.comhttps//xcap.ae.com:443>

But when I look at the wireshark capture that is not the URI that Snom is
sending.  I will do some more testing, but I was hoping that someone else
out there had a snom phone that could do some testing to confirm the same
issue.

On Wed, May 26, 2010 at 2:07 AM, Saúl Ibarra Corretgé [via OpenSIPS (Open
SIP Server)] 

> wrote:

> Hi,
>
> On 26/5/10 5:36 AM, osiris123d wrote:
>
> >
> > I just did a wireshark capture of a snom 8.2.29 phone sending a XCAP
> HTTPS
> > request to OpenXCAP and I think OpenXCAP is somehow mutating the resource
>
> > path.
> >
> >
> > The phone sends the correct path and all
> > 253 22:09:18.629127 192.168.0.7 173.203.87.134 HTTP GET
> >
> https://xcap.ae.com:443/xcap-r...@.../resource-lists/users/sip:9xx2xx2...@.../index<https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index>
>
> > HTTP/1.1
> >
> > But OpenXCAP replies with an error
> > 254 22:09:18.685811 173.203.87.134 192.168.0.7 HTTP HTTP/1.1 404 Not
> Found
> > (text/plain)
> >
> >
> > Here is what the OpenXCAP access.log says (You will notice that the first
>
> > two logs are the snom phone and after that is a successful access log of
> a
> > Counterpath Bria client)
> >
>
> Could you disable SSL and make a ngrep capture of the HTTP traffic? The
> fact that it worked with Bria but gives that error with Snom is
> surprising, I could tell you more by looking at a trace.
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

I just did a wireshark capture of a snom 8.2.29 phone sending a XCAP HTTPS
request to OpenXCAP and I think OpenXCAP is somehow mutating the resource
path.


The phone sends the correct path and all
253 22:09:18.629127 192.168.0.7 173.203.87.134  HTTPGET
https://xcap.ae.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/1.1 

But OpenXCAP replies with an error
254 22:09:18.685811 173.203.87.134  192.168.0.7 HTTPHTTP/1.1 404 
Not Found 
(text/plain)


Here is what the OpenXCAP access.log says (You will notice that the first
two logs are the snom phone and after that is a successful access log of a
Counterpath Bria client)

7X.XX.XXX.XXX 'GET
https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/
1.1' 404 0 179 'Mozilla/4.0 (compatible; snom360-SIP 8.2.29 1.1.3-m)' -
TRACEBACK (most recent call last):
  File
"/usr/lib/python2.5/site-packages/twisted/web2/channel/http.py", line 412,
in processRequest
self.request.process()
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 299, in process
d.callback(None)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 280, in callback
self._startRunCallbacks(result)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 354, in _startRunCallbacks
self._runCallbacks()
---  ---
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 371, in _runCallbacks
self.result = callback(self.result, *args, **kw)
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 296, in 
d.addCallback(lambda res, req: res.renderHTTP(req), self)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
335, in renderHTTP
d = self.authenticate(request)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
225, in authenticate
xcap_uri = parseNodeURI(uri, AuthenticationConfig.default_realm)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
94, in parseNodeURI
raise ResourceNotFound("XCAP root not found for URI: %s" %
node_uri)
xcap.errors.ResourceNotFound: XCAP root not found for URI:
https://xcap.ae.comhttps://xcap.aethercommunica
tions.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
7X.XX.XXX.XXX 'GET
https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
HTTP/
1.1' 404 0 179 'Mozilla/4.0 (compatible; snom360-SIP 8.2.29 1.1.3-m)' -
TRACEBACK (most recent call last):
  File
"/usr/lib/python2.5/site-packages/twisted/web2/channel/http.py", line 412,
in processRequest
self.request.process()
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 299, in process
d.callback(None)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 280, in callback
self._startRunCallbacks(result)
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 354, in _startRunCallbacks
self._runCallbacks()
---  ---
  File "/usr/lib/python2.5/site-packages/twisted/internet/defer.py",
line 371, in _runCallbacks
self.result = callback(self.result, *args, **kw)
  File "/usr/lib/python2.5/site-packages/twisted/web2/server.py",
line 296, in 
d.addCallback(lambda res, req: res.renderHTTP(req), self)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
335, in renderHTTP
d = self.authenticate(request)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
225, in authenticate
xcap_uri = parseNodeURI(uri, AuthenticationConfig.default_realm)
  File "/usr/lib/pymodules/python2.5/xcap/authentication.py", line
94, in parseNodeURI
raise ResourceNotFound("XCAP root not found for URI: %s" %
node_uri)
xcap.errors.ResourceNotFound: XCAP root not found for URI:
https://xcap.ae.comhttps://xcap.aethercommunica
tions.com:443/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/index
7X.XX.XXX.XXX 'GET
https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/contacts-re
source-list.xml HTTP/1.1' 401 0 141 - -
REQUEST headers:
Host: xcap.ae.com
Accept: */*
7X.XX.XXX.XXX 'GET
https://xcap.ae.com/xcap-r...@irock.com/resource-lists/users/sip:9xx2xx2...@irock.com/contacts-re
source-list.xml HTTP/1.1' 401 0 141 - -
REQUEST headers:
Host: xcap.ae.com
Accept: */*
Authorization: Basic c2lwOjkwMTI3MzIwMDlAaXJvY2suY29tOmIxNjdpbnV4
7X.XX.XXX.XXX 'GET
https://xcap.ae.com/xcap-r...@irock.com/org.openmobilealliance.pres-rules/users/sip:9xx2xx2...@ir
ock.com/pres-rules HTTP/1.1' 401 0 141 - -
REQUEST headers:
If-Non

[OpenSIPS-Users] bla_handle_notify() sends publish to wrong IP

[osiris123d]

I am trying to test and set up BLA and followed the following config
http://www.opensips.org/Resources/PuaBlaConfig

I am noticing that when my phone sends the notify and it executes the
bla_handle_notify() OpenSIPS is sending the publish to the actual domain
name IP address instead of the actual IP address thats in the location
table.  Here is an example

U 2010/05/25 20:01:23.568868 1XX.XXX.XXX.134:5060 -> 75.X.X.15X:2053
SUBSCRIBE sip:9xx2xx2...@75.x.x.15x:2053;line=293pwn59 SIP/2.0.
Via: SIP/2.0/UDP 1XX.XXX.XXX.134;branch=z9hG4bK2df4.72cce6b.0.
To: sip:9xx2xx2...@irock.com.
From: ;tag=144b207508c0e0296792f226e399eaf5-9670.
CSeq: 10 SUBSCRIBE.
Call-ID: 4959e3d95c1b26e4-18...@1xx.xxx.xxx.134.
Content-Length: 0.
User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)).
Max-Forwards: 70.
Event: dialog;sla.
Contact: .
Expires: 3610.
.

#
U 2010/05/25 20:01:23.639853 75.X.X.15X:2053 -> 1XX.XXX.XXX.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 1XX.XXX.XXX.134;branch=z9hG4bK2df4.72cce6b.0.
From: ;tag=144b207508c0e0296792f226e399eaf5-9670.
To: sip:9xx2xx2...@irock.com;tag=k87elwiitr.
Call-ID: 4959e3d95c1b26e4-18...@1xx.xxx.xxx.134.
CSeq: 10 SUBSCRIBE.
Contact: .
Expires: 3610.
Content-Length: 0.
.

#
U 2010/05/25 20:01:23.653515 75.X.X.15X:2053 -> 1XX.XXX.XXX.134:5060
NOTIFY sip:1XX.XXX.XXX.134:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.7:2053;branch=z9hG4bK-77nm5w2yoeob;rport.
From: sip:9xx2xx2...@irock.com;tag=k87elwiitr.
To: ;tag=144b207508c0e0296792f226e399eaf5-9670.
Call-ID: 4959e3d95c1b26e4-18...@1xx.xxx.xxx.134.
CSeq: 1 NOTIFY.
Max-Forwards: 70.
Contact: ;reg-id=1.
Event: dialog;sla.
Subscription-State: active.
Content-Type: application/dialog-info+xml.
Content-Length: 152.
.



#
U 2010/05/25 20:01:23.691668 1XX.XXX.XXX.134:5060 -> 75.X.X.15X:2053
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.7:2053;branch=z9hG4bK-c63fix17ewk4;rport=2053;received=75.X.X.15X.
From: "Blah Blah" ;tag=bnfufrbb7w.
To: "Blah Blah"
;tag=fa9ad7a35cf4c9d5a26f33f3220f505e.690f.
Call-ID: 3c26704f19b6-ih2vdtwqqd1u.
CSeq: 2 REGISTER.
Contact:
;q=0.5;expires=3600;received="sip:75.X.X.15X:2053".
Server: AE SIP Proxy.
Content-Length: 0.
.

#
U 2010/05/25 20:01:23.715677 1XX.XXX.XXX.134:5060 -> 97.74.144.17:5060
PUBLISH sip:9xx2xx2...@irock.com SIP/2.0.
Via: SIP/2.0/UDP 1XX.XXX.XXX.134;branch=z9hG4bK5d36.1de80852.0.
To: sip:9xx2xx2...@irock.com.
From: ;tag=144b207508c0e0296792f226e399eaf5-8f7a.
CSeq: 10 PUBLISH.
Call-ID: 4959e3d95c1b26e4-18...@1xx.xxx.xxx.134.
Content-Length: 139.
User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux)).
Max-Forwards: 70.
Event: dialog;sla.
Expires: 3601.
Content-Type: application/dialog-info+xml.
Sender: sip:9xx2xx2...@192.168.0.7:2053;line=293pwn59.
.





Since I am following the instructions from the link above what function am I
missing to make the publish get sent to the CONTACT or RECIEVED uri located
in the Location Database??
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Re: [OpenSIPS-Users] MySQL and multiple geographic servers

[osiris123d]

Will your clients be behind a firewall?  If so you would need to worry about
the natted devices not being able to recieve SIP messages from the secondary
SIP Proxy if the secondard ever needed to send SIP messages (example: Proxy
Primary is down, so Proxy Secondary takes over)
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Re: [OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

That's what I figured. Like Iñaki Baz Castillo said, XCAP doesn't seem to  
have a set standard. So I am not sure how Snom implemented XCAP. It does  
look like they are using xcapclient when you look at the logs.

On May 20, 2010 3:50pm, "Adrian Georgescu [via OpenSIPS (Open SIP Server)]"  
 wrote:




> I could not get it to work yet. Is mystery meat



> Adrian



> On May 20, 2010, at 8:03 PM, osiris123d wrote:



> >


> > There has been a separate thread about "SIP Presence Aggregation" and I  
> was


> > just wondering if the people who have gotten Snom phones to work with


> > OpenXCAP and presence could share the Settings Copy and Paste of the  
> phone.


> >


> > I have messed with the phones to no avail. The Snom Forum isn't very  
> active


> > with answers. When I set up the phone to talk to the XCAP server I see  
> it


> > communicating but for all I know it might not be getting past the https


> > encryption. I imported the certificate, but that didn't help things.


> >


> > Any info would be appreciated if you have any.


> > --


> > View this message in context:  
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> >


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> >




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[OpenSIPS-Users] OpenXCAP and Presence with SNOM version 8.2.29

[osiris123d]

There has been a separate thread about "SIP Presence Aggregation" and I was
just wondering if the people who have gotten Snom phones to work with
OpenXCAP and presence could share the Settings Copy and Paste of the phone.

I have messed with the phones to no avail.  The Snom Forum isn't very active
with answers.  When I set up the phone to talk to the XCAP server I see it
communicating but for all I know it might not be getting past the https
encryption.  I imported the certificate, but that didn't help things.

Any info would be appreciated if you have any.
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Re: [OpenSIPS-Users] SIP Presence Aggregation Issue

[osiris123d]

Is XCAP really a good option?  It works well with Counterpath Bria softphone
clients, but other then that I don't believe there are any hard phones out
there that support it.

Snom's v8 software says that it is XCAP capable, but I have yet to get it
working with OpenXCAP.

So if no phones support XCAP why would it even be an option.
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Re: [OpenSIPS-Users] Private IP in registered AOR causing failure

[osiris123d]

opensipslist,

I am not sure if you ever figured this out or not.  I just ran into the same
issue and the way to resolve it is to do the following before
handle_subscribe()

fix_nated_register()


So you would need to do


fix_nated_register()
handle_subscribe()


I hope that is the issue you were having.
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Re: [OpenSIPS-Users] Stun Module - OpenSIPS won't start

[osiris123d]

Thanks. Just looked at RFC 3489 and didn't know STUN had to have a second  
IP. Makes sense why it needs one now.

On May 7, 2010 2:56am, "Bogdan-Andrei Iancu [via OpenSIPS (Open SIP  
Server)]"  wrote:




> Hi,



> you need to define the secondary IP and port params too.



> Regards,


> Bogdan



> osiris123d wrote:


> > I just compiled the STUN module on Debian 5.0.4 x64 and added the  
> following


> > stun parameters to my config


> >


> >


> > 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:173.*.*.134:5060


> >


> >


> > #**


> > # - stun params -


> > modparam("stun","primary_ip","173.*.*.134")


> > modparam("stun","primary_port","5060")


> >


> >


> >


> >


> > When I try and start OpenSIPS it fails and I see the following errors in


> > syslog. Any idea whats up?


> >


> >


> >


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:  
> DBG:core:init_mod:


> > initializing module stun


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if host==us: 14==14 && [173.*.*.134]


> > == [173.*.*.134]


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if port 5060 matches port 5060


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if host==us: 14==14 && [173.*.*.134]


> > == [173.*.*.134]


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if port 5060 matches port 3478


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:stun:stun_mod_init: socketfd2 not found


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if host==us: 13==14 &&  
> [192.168.2.143] =


> > = [173.*.*.134]


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if port 5060 matches port 5060


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:stun:stun_mod_init: socketfd3 not found


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if host==us: 13==14 &&  
> [192.168.2.143] =


> > = [173.*.*.134]


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:core:grep_sock_info: checking if port 5060 matches port 3478


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:stun:stun_mod_init: socketfd4 not found


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > DBG:stun:stun_mod_init: stun init failed


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:


> > ERROR:core:init_mod: failed to initialize module stun


> > May 6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: ERROR:core:main:


> > error while initializing modules


> >




> --


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> www.voice-system.ro




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[OpenSIPS-Users] Stun Module - OpenSIPS won't start

[osiris123d]

I just compiled the STUN module on Debian 5.0.4 x64 and added the following
stun parameters to my config


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:173.*.*.134:5060  


#**
# - stun params -
modparam("stun","primary_ip","173.*.*.134")
modparam("stun","primary_port","5060")




When I try and start OpenSIPS it fails and I see the following errors in
syslog.  Any idea whats up?



May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: DBG:core:init_mod:
initializing module stun
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if host==us: 14==14 &&  [173.*.*.134] 
== [173.*.*.134]
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if port 5060 matches port 5060
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if host==us: 14==14 &&  [173.*.*.134] 
== [173.*.*.134]
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if port 5060 matches port 3478
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:stun:stun_mod_init: socketfd2 not found
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if host==us: 13==14 &&  [192.168.2.143] =
= [173.*.*.134]
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if port 5060 matches port 5060
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:stun:stun_mod_init: socketfd3 not found
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if host==us: 13==14 &&  [192.168.2.143] =
= [173.*.*.134]
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:core:grep_sock_info: checking if port 5060 matches port 3478
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:stun:stun_mod_init: socketfd4 not found
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
DBG:stun:stun_mod_init: stun init failed
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]:
ERROR:core:init_mod: failed to initialize module stun
May  6 22:53:38 Proxy01 /usr/local/sbin/opensips[11212]: ERROR:core:main:
error while initializing modules
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Re: [OpenSIPS-Users] Presence - Watchers table question

[osiris123d]

Thats what I was kind of thinking but wasn't sure. I see the distinction  
between delete and block, but wanted to verify.

Thanks

On May 5, 2010 3:16am, "Anca Vamanu-2 [via OpenSIPS (Open SIP Server)]"  
 wrote:




> osiris123d wrote:


> > I have posted this email once before but it was blocked due to being  
> greater


> > than 40KB. So if it finally does get approved and sent out I apologies  
> for


> > it being sent twice.


> >


> > I am wondering if something is not right here. Here is the scenario  
> (Using


> > two Counterpath Bria clients)


> >


> > Client A adds Client B to its contact list


> > Client B recieves a request and he accepts the request


> > Now both Client A and Client B can see each others presence


> > (All good so far)


> >


> > Now Client A deletes Client B from his contact list


> > Client B only sees Client A's presence as Offline


> > (All good so far)


> >


> > Now Client A adds Client B back to its contact list


> > (Here is where things don't seem to work right)


> > Client B doesn't recieve a request at all, but Client A can now see  
> Client


> > B's presence.


> >


> >


> >


> > Is this how Presence rules are suppose to work? it would seem to me  
> that in


> > the second part of my scenario when Client A deletes Client B, the  
> record in


> > the Watchers table should be deleted. That way if the user ever wanted  
> to


> > add that person back they would once again receive a friend request and  
> have


> > the option to accept or decline.


> >


> > Am I wrong here?


> >


> Hi,



> In fact it does seem correct to me. Deleting a contact from the buddy


> list and denying someone to see your presence status are two different


> things.


> Why if client A deleted contact B from his buddy list did you expect


> client A to loose its permission to see B's state? There is no


> connection between the two. For client A to be denied to see B's state,


> B is the one that must specify this..


> Anyhow this question is more related to the clients, and the way they


> correlate deleting contacts with presence rules, opensips will do what


> the client tells it. And presence rules are specified in pres-rules XCAP


> documents. So if you want to report something not working in opensips


> you should look after inconsistencies between pres-rules documents and


> states in watchers tables.



> Regards,


> Anca



> --


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> www.voice-system.ro




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[OpenSIPS-Users] Presence - Watchers table question

[osiris123d]

I have posted this email once before but it was blocked due to being greater
than 40KB.  So if it finally does get approved and sent out I apologies for
it being sent twice.

I am wondering if something is not right here.  Here is the scenario (Using
two Counterpath Bria clients)

Client A adds Client B to its contact list 
Client B recieves a request and he accepts the request 
Now both Client A and Client B can see each others presence 
(All good so far) 

Now Client A deletes Client B from his contact list 
Client B only sees Client A's presence as Offline 
(All good so far) 

Now Client A adds Client B back to its contact list 
(Here is where things don't seem to work right) 
Client B doesn't recieve a request at all, but Client A can now see Client
B's presence. 



Is this how Presence rules are suppose to work?  it would seem to me that in
the second part of my scenario when Client A deletes Client B, the record in
the Watchers table should be deleted.  That way if the user ever wanted to
add that person back they would once again receive a friend request and have
the option to accept or decline.

Am I wrong here?
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[OpenSIPS-Users] Opensips-mi-proxy xcap-diff issues

[osiris123d]

I have OpenXCAP, OpenSIPS and Opensips-mi-proxy all installed on one machine. 
For the most part things are working except for when I enable xcap-diff in
the OpenXCAP config.ini (enable_publish_xcapdiff = yes).

When it's enabled the xcap-diff calls to OpenSIPS-MI-Proxy blow up.  The
logs below are from me deleting a buddy contact from Counterpath's Bria
client.  

Here is the syslog

Apr 30 10:55:00 Proxy01 openxcap[31738]: Starting factory

Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: Got XMLRPC request from
173.x.x.x: refreshWatchers (sip:901273x...@coolbeans.c
om presence 0)
Apr 30 10:55:00 Proxy01 /usr/local/sbin/opensips[17800]:
INFO:presence:send_notify_request: NOTIFY sip:901273x...@irock.com via sip:
901273x...@192.251.125.224:61877;transport=udp on behalf of
sip:901273x...@coolbeans.com for event presence
Apr 30 10:55:00 Proxy01 /usr/local/sbin/opensips[17800]:
INFO:presence:send_notify_request: NOTIFY sip:901273x...@irock.com via sip:
901273x...@192.251.125.224:61877;transport=udp on behalf of
sip:901273x...@irock.com for event presence
Apr 30 10:55:00 Proxy01 /usr/local/sbin/opensips[17800]:
INFO:presence:send_notify_request: NOTIFY sip:901273x...@coolbeans.com via 
sip:901273x...@192.251.125.224:12966;transport=udp on behalf of
sip:901273x...@coolbeans.com for event presence.winfo
Apr 30 10:55:00 Proxy01 openxcap[31738]: Starting factory

Apr 30 10:55:00 Proxy01 openxcap[31738]: Stopping factory

Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: Got XMLRPC request from
173.x.x.x: pua_publish (sip:901273x...@coolbeans.com 3
600 xcap-diff application/xcap-diff+xml . . 
Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: http://xcap.AE.com/xcap-root";>
Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: http://xcap.aethercommun
ications.com/xcap-root/org.openmobilealliance.pres-rules/users/901273/pres-rules"
previous-etag="8358dd69589d5333238f13b08fa110b
c"/>
Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: 
Apr 30 10:55:00 Proxy01 opensips-mi-proxy[2645]: )

 
Apr 30 10:55:01 Proxy01 openxcap[31738]: Starting factory

Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: Got XMLRPC request from
173.x.x.x: pua_publish (sip:901273x...@coolbeans.com 3
600 xcap-diff application/xcap-diff+xml . . 
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: http://xcap.AE.com/xcap-root";>
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: http://xcap.AE.com/xcap-root/resource-lists/users/901273/resource-list.xml";
previous-etag="caf3949208bb0a57935845ae98aaae14"/>
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: 
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: )
Apr 30 10:55:01 Proxy01 openxcap[31738]: Starting factory

Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: Got XMLRPC request from
173.x.x.x: pua_publish (sip:901273x...@coolbeans.com 3
600 xcap-diff application/xcap-diff+xml . . 
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: http://xcap.AE.com/xcap-root";>
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: http://xcap.AE.com/xcap-root/resource-lists/users/901273/contacts-resource-list.xml";
previous-etag="95660297b918e64efbc2c9a80b05ed78"/
>
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: 
Apr 30 10:55:01 Proxy01 opensips-mi-proxy[2645]: )


Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: error: Error while
processing command pua_publish (sip:901273x...@coolbeans.com 360
0 xcap-diff application/xcap-diff+xml . . 
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: error: http://xcap.AE.com/xcap-root";>
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: error: http://xcap.AE.com/xcap-root/org.openmobilealliance.pres-rules/users/901273/pres-rules";
previous-etag="8358dd69589d5333238f13b0
8fa110bc"/>
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: error: 
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: error: ): OpenSIPS command
did timeout
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: Exception rendering:
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: Traceback (most recent call
last):
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]:   File
"/usr/lib/python2.5/site-packages/twisted/internet/defer.py", line 371, in _
runCallbacks
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: self.result =
callback(self.result, *args, **kw)
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]:   File
"/usr/lib/pymodules/python2.5/miproxy/proxy.py", line 88, in _cbRender
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: code = int(code)
Apr 30 10:55:03 Proxy01 opensips-mi-proxy[2645]: ValueError: invalid literal
for int() with base 10: 'OpenSIPS'
Apr 30 10:55:03 Proxy01 openxcap[31738]: error: Unhandled error in Deferred:
Apr 30 10:55:03 Proxy01 openxcap[31738]: ValueError: ('500', 'Internal
Server Error')
Apr 30 10:55:03 Proxy01 openxcap[31738]: Stopping factory

Apr 30 10:55:04 Proxy01 opensips-mi-proxy[2645]: error: Error while
processing command pua_publish (sip:901273x...@coolbeans.com 360
0 xcap-diff application/xcap-diff+xml . . 
Apr 30 10:55:04 Proxy01 opensips-mi-proxy[2645]: error: http://xcap.AE

Re: [OpenSIPS-Users] SIPSimple SDK on Windows

[osiris123d]

Thanks for the reply.

On Apr 23, 2010 11:14am, "Adrian Georgescu [via OpenSIPS (Open SIP  
Server)]"  wrote:




> We are currently working on porting the SDK to MS Windows. No ETA yet


> though but it will be fully documented.



> Adrian



> On Apr 23, 2010, at 6:06 PM, osiris123d wrote:



> >


> > Just wondering if anyone has had any success installing SIPSimple on


> > Windows?


> > I know that the http://sipsimpleclient.com/wiki/SipInstallation


> > SIPSIMPLE


> > homepage says that its been tested and supported on Linux and OSX,


> > but I


> > just thought I would try.


> >


> > I installed windows GNU, patch, python, Cython, svn, MinGW and


> > Msys. I go


> > into the python-sipsimple-0.14.2 folder and execute "python setup.py


> > install" and everything starts out good, but I get the following error


> >


> > Fetching PJSIP from SVN repository


> > Using SVN revision 2830


> > Patching PJSIP


> > error: Got return value 1 while executing "patch --forward -d


> > build\temp.win32-2.6\Release\pjsip -p0 -i


> > c:\python-sipsimple-0.14.2\patches


> > \sdp_neg_cancel_remote_offer_r2669.patch


> >


> > Just to see whats going on I try to execute this exact command on my


> > own and


> > I see the following


> >


> > patching file pjmedia/include/pjmedia/sdp_neg.h


> > missing header for unified diff at line 36 of patch


> > can't find file to patch at input line 36


> > Perhaps you used the wrong -p or --strip option?


> > The text leading up to this was:


> > --


> > | * create_w_remote_offer() +--+


> > | *


> > | *


> > --


> > File to patch:


> >


> >


> >


> >


> > For the record I am able to install pjsip successfully on the windows


> > machine using the source from the pjsip webpage.


> >


> > This isn'ta big deal. I just wanted to try and play with the


> > sipsimple sdk


> > and try and make a windows app.


> >


> >


> > --


> > View this message in context:  
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> > Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


> >


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> >




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[OpenSIPS-Users] SIPSimple SDK on Windows

[osiris123d]

Just wondering if anyone has had any success installing SIPSimple on Windows? 
I know that the  http://sipsimpleclient.com/wiki/SipInstallation SIPSIMPLE
homepage  says that its been tested and supported on Linux and OSX, but I
just thought I would try.

I installed windows GNU, patch, python, Cython, svn, MinGW and Msys.  I go
into the python-sipsimple-0.14.2 folder and execute "python setup.py
install" and everything starts out good, but I get the following error

Fetching PJSIP from SVN repository
Using SVN revision 2830
Patching PJSIP
error: Got return value 1 while executing "patch --forward -d
build\temp.win32-2.6\Release\pjsip -p0 -i
c:\python-sipsimple-0.14.2\patches\sdp_neg_cancel_remote_offer_r2669.patch

Just to see whats going on I try to execute this exact command on my own and
I see the following

patching file pjmedia/include/pjmedia/sdp_neg.h
missing header for unified diff at line 36 of patch
can't find file to patch at input line 36
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
| * create_w_remote_offer() +--+
|  *
|  * 
--
File to patch:




For the record I am able to install pjsip successfully on the windows
machine using the source from the pjsip webpage.

This isn't a big deal.  I just wanted to try and play with the sipsimple sdk
and try and make a windows app.


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[OpenSIPS-Users] OpenSIPS with MySQL Cluster NDBCLUSTER

[osiris123d]

Has anyone run into any issue with using MySQL Cluster 7.x with OpenSIPS
database?  I know the default mysql engine for opensips is MyISAM and 
http://n2.nabble.com/mysql-engine-type-for-opensips-tables-td3666233.html#a3666233
Jeff Pyle  seems to be using it.  Just wondering if anyone had any
experience with it.

Was also wondering if it would be possible to have a feature request to
implement the NDB API within OpenSIPS so that you wouldn't need the MySQL
App Nodes?  Wasn't sure if that would be very helpful to anyone.

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Re: [OpenSIPS-Users] DB-ONLY mode call between two proxies failed.

[osiris123d]

Makes sense.  Thanks

On Fri, Mar 26, 2010 at 5:32 AM, Bogdan-Andrei Iancu [via OpenSIPS (Open SIP
Server)] 

> wrote:

>
> osiris123d wrote:
> > Bogdan,
> >
> > Are you saying that its not possible to have two Active/Active SIP
> Proxies
> > share the same Location table when the clients are NATed?
> >
> as the servers have different IPs, yes, it is not possible.
> > I currently have two OpenSIPS proxies set up and sharing the same
> database
> > and if NATed Client A registered with Proxy1 and NATed Client B
> registered
> > with Proxy2, then when Client A calls Client B the call fails.  Is it
> > because of the two Active/Active Proxies and the NATed clients?
> >
> > When I ngrep I see that the foreign proxy is sending the sip invite to
> the
> > correct public IP:rport, but internally behind the firewall no traffic is
>
> > reaching the client.
> >
> yes, because the nat pinhole is open only towards/from Proxy1 (where the
> nated client registered), so proxy2 will not be able to cross that nat.
>
> Regards,
> Bogdan
>
>
> --
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> www.voice-system.ro
>
>
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Re: [OpenSIPS-Users] DB-ONLY mode call between two proxies failed.

[osiris123d]

My question was referencing this email thread


Hi Jiang, 

the problem is the USRLOC module saves (in the user location record) the 
socket the REGISTER was received from (to use it when sending the 
traffic to the users). But if you have multiple servers on the same DB, 
there will be a kind of conflict between the stored sockets (record was 
inserted by R1 with socket S1 and later R2 reads the records, R2 will 
try to use S1 which obviously is not local). 

I can send you a small patch do get rid of that, but if you are dealing 
with NATed clients, it will not works - a natted client can be handled 
only by the same server. 

Regards, 
Bogdan 

Jiang Jinke wrote: 

> I'm using the same location table with two/three proxy server, but I 
> can't make calls between the proxies. 
> Do I have to do replication in SIP level ? I'd like to do this in a 
> simple way, but if replication will make the arch more flexible, then 
> it's acceptable with a more complicate config. 
> 
> I'm having the following message in my syslog when the other endpoint 
> is not in the same proxy server which was receiving the call. 
> 
> syslog content: 
> WARNING:usrloc:dbrow2info: non-local socket
> ...ignoring 
> ERROR:core:get_out_socket: no socket found 
> ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 1  (no 
> corresponding listening socket) 
> ERROR:tm:t_forward_nonack: failure to add branches 
> 
> Regards, 
> Jinke Jiang 

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Re: [OpenSIPS-Users] DB-ONLY mode call between two proxies failed.

[osiris123d]

Bogdan,

Are you saying that its not possible to have two Active/Active SIP Proxies
share the same Location table when the clients are NATed?

I currently have two OpenSIPS proxies set up and sharing the same database
and if NATed Client A registered with Proxy1 and NATed Client B registered
with Proxy2, then when Client A calls Client B the call fails.  Is it
because of the two Active/Active Proxies and the NATed clients?

When I ngrep I see that the foreign proxy is sending the sip invite to the
correct public IP:rport, but internally behind the firewall no traffic is
reaching the client.
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[OpenSIPS-Users] Call Issues with MySQL Latency

[osiris123d]

Wondering if anyone has ran into an issue like this

I have a server with a public IP 66.X.X.X that is running
OpenSIPSMediaDispatcher and also MySQL.  I set up a second
OpenSIPS/MediaDispatcher with a public IP of 12.X.X.X and pointed the config
to the 66.X.X.X for all database queries.  I also have a MediaProxy-Relay
server at 66.X.X.X

When I point phones to the first OpenSIPS server and make calls everything
is fine.  When I point phones to the second OpenSIPS server I get weird
issues.  The caller makes an INVITE request and is asked to authenticate and
it does.  After that I think there is a very small delay so the phone thinks
nothing happened so it sends the INVITE again.  The Callee phone ends up
getting multiple calls coming in and things start to error out.  

The ping time from the second OpenSIPS server to the first opensips server
with the mysql database on it is about 83 ms.  Could that be the issue? 
Should an OpenSIPS server always have low latency between it and the
database?  I am going to try and get another OpenSIPS server up on the
66.X.X.X network and see if that fixes things, but wanted to know if this
could be causing my issues.

Here is a trace of what is happening.

U 2010/03/25 16:33:48.115931 192.251.125.224:8500 -> 12.X.X.X:5060
INVITE sip:2732...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-2aea8dec357268f6-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: .
From: "Test irock";tag=c29eb309.
Call-ID: MTBlOWMyYmU2ZWIzNjcwNjM3ZGY4NGY5MzdlMzNlZjg..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 185.
.
v=0.
o=- 4 2 IN IP4 192.168.100.82.
s=CounterPath Bria.
c=IN IP4 192.168.100.82.
t=0 0.
m=audio 10472 RTP/AVP 9 0 3 101.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2010/03/25 16:33:48.116897 12.X.X.X:5060 -> 192.251.125.224:8500
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-2aea8dec357268f6-1---d8754z-;rport=8500;received=192.251.125.224.
To: ;tag=b7648717ba368896b55480d883b679b5.f6f1.
From: "Test irock";tag=c29eb309.
Call-ID: MTBlOWMyYmU2ZWIzNjcwNjM3ZGY4NGY5MzdlMzNlZjg..
CSeq: 1 INVITE.
Proxy-Authenticate: Digest realm="irock.com",
nonce="4babd6da0022d5aefc2125240a3650bd1657722ce687", qop="auth".
Server: AC SIP Proxy.
Content-Length: 0.
.

#
U 2010/03/25 16:33:48.209391 192.251.125.224:8500 -> 12.X.X.X:5060
ACK sip:2732...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-2aea8dec357268f6-1---d8754z-;rport.
Max-Forwards: 70.
To: ;tag=b7648717ba368896b55480d883b679b5.f6f1.
From: "Test irock";tag=c29eb309.
Call-ID: MTBlOWMyYmU2ZWIzNjcwNjM3ZGY4NGY5MzdlMzNlZjg..
CSeq: 1 ACK.
Content-Length: 0.
.

#
U 2010/03/25 16:33:48.215864 192.251.125.224:8500 -> 12.X.X.X:5060
INVITE sip:2732...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-ea8f95b806a53ac7-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: .
From: "Test irock";tag=c29eb309.
Call-ID: MTBlOWMyYmU2ZWIzNjcwNjM3ZGY4NGY5MzdlMzNlZjg..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
Proxy-Authorization: Digest
username="9XX2731999",realm="irock.com",nonce="4babd6da0022d5aefc2125240a3650bd1657722ce687",uri="sip:2732...@irock.com",response="8585744c9a24c899f9209437cd8f0497",cnonce="40a57a2871e80cd90d93ce34784fd701",nc=0001,qop=auth,algorithm=MD5.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 185.
.
v=0.
o=- 4 2 IN IP4 192.168.100.82.
s=CounterPath Bria.
c=IN IP4 192.168.100.82.
t=0 0.
m=audio 10472 RTP/AVP 9 0 3 101.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2010/03/25 16:33:48.704814 192.251.125.224:8500 -> 12.X.X.X:5060
INVITE sip:2732...@irock.com SIP/2.0.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-ea8f95b806a53ac7-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: .
From: "Test irock";tag=c29eb309.
Call-ID: MTBlOWMyYmU2ZWIzNjcwNjM3ZGY4NGY5MzdlMzNlZjg..
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO.
Content-Type: application/sdp.
Proxy-Authorization: Digest
username="9XX2731999",realm="irock.com",nonce="4babd6da0022d5aefc2125240a3650bd1657722ce687",uri="sip:2732...@irock.com",response="8585744c9a24c899f9209437cd8f0497",cnonce="40a57a2871e80cd90d93ce34784fd701",nc=0001,qop=auth,algorithm=MD5.
User-Agent: Bria release 2.5.4 stamp 53956.
Content-Length: 185.
.
v=0.
o=- 4 2 IN IP4 192.168.100.82.
s=CounterPath Bria.
c=IN IP4 192.168.100.82.
t=0 0.
m=audio 10472 RTP/AVP 9 0 3 101.
a=sendrecv.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 2010/03/25 16:33:48.800403 12.X.X.X:5060 -> 192.251.125.224:8500
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.100.82:27038;branch=z9hG4bK-d8754z-