[OpenSIPS-Users] opensips 1.8 tls version

[Engineer voip]

Hello all,

where i get the opensips 1.8 tls versions please?
-- 

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Re: [OpenSIPS-Users] 403 reply forbidden

[yassine.ab]

Hi Bogdan,

Here my log for opensips

I have always error 403 Preload Route denied .

Thanks in advance

regards
yassine

Message du 27/08/12 16:16
De : "Bogdan-Andrei Iancu" 
A : "OpenSIPS users mailling list" 
Copie à : "yassine.ab" 
Objet : Re: [OpenSIPS-Users] 403 reply forbidden

Hi Yassine,

How the original INVITE looks like, especially the Route header ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/24/2012 10:55 PM, yassine.ab wrote:
> Hi,
>
> thank you for your answer,
> I have already tested this option, now I have another error, 403 preload
> route denied , looking a bit on the forum, I found this post
>
> *http://opensips-open-sip-server.1449251.n2.nabble.com/Registration-and-Loose-Route-td3178163.html*
>
> So up to now , I added the alias in opensips.cfg,
> *alias="open-ims.test"*
>
> I added a domain in the table domain using *opensipsctl domain add domain*,
>
> I added also in opensips.cfg that:
>
> *loadmodule "domain.so"
> # - domain params -
> modparam("domain", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips")
> modparam("domain", "db_mode", 1) # Use caching*
>
> but no results
>
> regards
> yassine
>
>
>
> --
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/403-reply-forbidden-tp7581343p7581400.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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>


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Quand Jean-Luc Delarue parlait de son grand amour… à lire sur Voila.fr 
http://people.voila.fr/people/actu-stars/personnalites/quand-jean-luc-delarue-parlait-d-anissa-kehl-son-grand-amour-people_8397.htmlAug 31 11:17:06 [5758] DBG:core:parse_msg: SIP Request:
Aug 31 11:17:06 [5758] DBG:core:parse_msg:  method:  
Aug 31 11:17:06 [5758] DBG:core:parse_msg:  uri: 
Aug 31 11:17:06 [5758] DBG:core:parse_msg:  version: 
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=2
Aug 31 11:17:06 [5758] DBG:core:parse_via_param: found param type 232,  
= ; state=16
Aug 31 11:17:06 [5758] DBG:core:parse_via: end of header reached, state=5
Aug 31 11:17:06 [5758] DBG:core:parse_headers: via found, flags=2
Aug 31 11:17:06 [5758] DBG:core:parse_headers: this is the first via
Aug 31 11:17:06 [5758] DBG:core:receive_msg: After parse_msg...
Aug 31 11:17:06 [5758] DBG:core:receive_msg: preparing to run routing scripts...
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=100
Aug 31 11:17:06 [5758] DBG:core:parse_to: end of header reached, state=9
Aug 31 11:17:06 [5758] DBG:core:parse_to: display={}, 
ruri={sip:t...@open-ims.test}
Aug 31 11:17:06 [5758] DBG:core:get_hdr_field:  [24]; 
uri=[sip:t...@open-ims.test] 
Aug 31 11:17:06 [5758] DBG:core:get_hdr_field: to body [sip:t...@open-ims.test
]
Aug 31 11:17:06 [5758] DBG:core:get_hdr_field: cseq : <10> 
Aug 31 11:17:06 [5758] DBG:core:get_hdr_field: content_length=0
Aug 31 11:17:06 [5758] DBG:maxfwd:is_maxfwd_present: value = 10 
Aug 31 11:17:06 [5758] DBG:uri:has_totag: no totag
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=78
Aug 31 11:17:06 [5758] DBG:tm:t_lookup_request: start searching: hash=43782, 
isACK=0
Aug 31 11:17:06 [5758] DBG:tm:matching_3261: RFC3261 transaction matching failed
Aug 31 11:17:06 [5758] DBG:tm:t_lookup_request: no transaction found
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=200
Aug 31 11:17:06 [5758] DBG:core:get_hdr_field: found end of header
Aug 31 11:17:06 [5758] DBG:rr:find_first_route: No Route headers found
Aug 31 11:17:06 [5758] DBG:rr:loose_route: There is no Route HF
Aug 31 11:17:06 [5758] DBG:core:grep_sock_info: checking if host==us: 12==12 && 
 [192.168.0.70] == [192.168.0.70]
Aug 31 11:17:06 [5758] DBG:core:grep_sock_info: checking if port 5060 matches 
port 5060
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=800
Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=
Aug 31 11:17:06 [5758] DBG:usrloc:update_ucontact: exists callback for type= 
UL_CONTACT_UPDATE
Aug 31 11:17:06 [5758] DBG:registrar:build_contact: created Contact HF: 
Contact: ;expires=600120

Aug 31 11:17:06 [5758] DBG:core:parse_headers: flags=
Aug 31 11:17:06 [5758] DBG:core:check_ip_address: params 192.168.0.72, 
192.168.0.72, 0
Aug 31 11:17:06 [5758] DBG:core:destroy_avp_list: destroying list (nil)
Aug 31 11:17:06 [5758] DBG:core:receive_msg: cleaning up
Aug 31 11:17:12 [5758] DBG:core:parse_msg: SIP Request:
Aug 31 11:17:12 [5758] DBG:core:parse_msg:  method:  
Aug 31 11:17:12 [5758] DBG:core:parse_msg:  uri: 
Aug 31 11:17:12 [5758] DBG:core:parse_msg:  version: 
Aug 31 11:17:12 [5758] DBG:core:parse_headers: flags=2
Aug 31 11:17:12 [5758] DBG:core:parse_via_param: found para

Re: [OpenSIPS-Users] opensips 1.8 tls version

[Muhammad Shahzad]

You need to enable it at compile time.

Thank you.


On Fri, Aug 31, 2012 at 1:27 PM, Engineer voip  wrote:

> Hello all,
>
> where i get the opensips 1.8 tls versions please?
> --
>
> Best Regards
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
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---
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Cell: +92 334 422 40 88
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Re: [OpenSIPS-Users] opensips 1.8 tls version

[Engineer voip]

Hi Muhammad,

I already  installed my opensips, what can i do to recompile my opensips to
have TLS option ?



2012/8/31 Muhammad Shahzad 

> You need to enable it at compile time.
>
> Thank you.
>
>
> On Fri, Aug 31, 2012 at 1:27 PM, Engineer voip  wrote:
>
>> Hello all,
>>
>> where i get the opensips 1.8 tls versions please?
>> --
>>
>> Best Regards
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


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Re: [OpenSIPS-Users] 403 reply forbidden

[Bogdan-Andrei Iancu]

Once again, post here the SIP INVITE requests that is rejected by 
OpenSIPS (use ngrep for capturing).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/31/2012 12:37 PM, yassine.ab wrote:

Hi Bogdan,

Here my log for opensips

I have always error 403 Preload Route denied .

Thanks in advance

regards
yassine

Message du 27/08/12 16:16
De : "Bogdan-Andrei Iancu"
A : "OpenSIPS users mailling list"
Copie à : "yassine.ab"
Objet : Re: [OpenSIPS-Users] 403 reply forbidden

Hi Yassine,

How the original INVITE looks like, especially the Route header ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/24/2012 10:55 PM, yassine.ab wrote:

Hi,

thank you for your answer,
I have already tested this option, now I have another error, 403 preload
route denied , looking a bit on the forum, I found this post

*http://opensips-open-sip-server.1449251.n2.nabble.com/Registration-and-Loose-Route-td3178163.html*

So up to now , I added the alias in opensips.cfg,
*alias="open-ims.test"*

I added a domain in the table domain using *opensipsctl domain add domain*,

I added also in opensips.cfg that:

*loadmodule "domain.so"
# - domain params -
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("domain", "db_mode", 1) # Use caching*

but no results

regards
yassine



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Re: [OpenSIPS-Users] opensips 1.8 tls version

[Muhammad Shahzad]

Dear, like every opensource software, opensips also comes with some
standard files like LICENSE, INSTALL, README, CHANGELOG etc. etc. It seem
you are very new to opensource software and technologies, so i suggest you
start with reading them first, specially INSTALL file, it will tell you
exactly how opensips is installed, what are its dependencies and what
install options are available etc. etc.

All the questions i saw from you on mailing list, have already been asked
by so many people, so many times and have been answered by so many members
of list. I don't want to be rude here but you seriously need to improve
your search skills which will save you and others from frustration and
wastage of time.

For OpenSIPs, you can start from here,

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

On the left pane under Resources you will find many useful links which will
answer most common questions you may have in your mind. Try reading those
links instead of flooding mailing list with most common and obvious
questions. Only ask community for questions which are either too complex to
understand or too specific to your setup.

Hope this helps.

Thank you.


On Fri, Aug 31, 2012 at 2:20 PM, Engineer voip  wrote:

> Hi Muhammad,
>
> I already  installed my opensips, what can i do to recompile my opensips
> to have TLS option ?
>
>
>
>
> 2012/8/31 Muhammad Shahzad 
>
>> You need to enable it at compile time.
>>
>> Thank you.
>>
>>
>> On Fri, Aug 31, 2012 at 1:27 PM, Engineer voip wrote:
>>
>>> Hello all,
>>>
>>> where i get the opensips 1.8 tls versions please?
>>> --
>>>
>>> Best Regards
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>> --
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
>
> Best Regards
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [OpenSIPS-Users] A Dynamic List of Gateways

[Brett Nemeroff]

What kind of traffic are you expecting? In CPS?

It may not be an issue.
-Brett


On Thu, Aug 30, 2012 at 9:55 PM, Shaahin Madani wrote:

> Greetings,
>
> Thanks for the helpful replies. Presently, the list of gateways cannot be
> retrieved using a direct query on the external database, and hence I need
> the Perl script to act as the middle-ware too. Some changes in the external
> database may fix this, and then I would be able to use the avops module
> as Ali suggested, but I need to first thoroughly study all the
> consequences ...
>
> Regarding the bottleneck issue, I don't see any way that I can avoid the
> queries (or Perl). In case it turns out to be a real bottleneck, then is
> there any way to go around it, or will I actually be hitting an OpenSIPS
> limitation?
>
>
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Re: [OpenSIPS-Users] Parallel forking and per branch manipulation

[Jayesh Nambiar]

Hi Binan,
Thank you very much for your reply. What you wrote is what I understand
from all the documentation present but probably the concept is not working
for me or probably just not getting into my head. This is a snippet of what
I'm trying to do:

lookup("location");
$branch = $ru;

avp_db_query("SELECT cli, phone_number FROM users WHERE user_id = '$rU',
"$avp(cli), $avp(parallel_number)");
$rU = $avp(parallel_number);
$rd = my_gateway_ip;
$var(uri) = "sip:"+$rU+"@"+$rd;

$branch = $var(ruri);

t_on_branch("1"); # --This is confusing me---
t_relay();
exit;

branch_route[1] { #---Need to know how to check for branch id here---
   if(branch_id == 1) { #--- How do I check for this condition??
  uac_replace_from("sip:$avp(cli)@mydomain.com");
   }
}

The problem is I dont know how to identify the branch id in the branch
route. Even if I set branch flag for second branch using setbflag(1,1) and
I check for this flag in the branch route using isbflagset(1,1), by the
time the first branch goes into branch_route, the second branch is created
and the condition isbflagset(1,1) becomes true and the uac_replace_from()
gets executed for first branch also. My requirement is to execute this
branch_route only for second branch which is branch_id 1.
Isn't there any straight forward way of checking the id of the branch when
it comes to the branch route??

Any help is really appreciated. Thanks.

--- Jayesh

> Hi,
>
> lets say you have 2 branches 0 towards user in location and 1 towards PSTN.
> ?
> I suggest to use branch flags which are used to activate some functions in
> branch level
>
> when creating the branches in main route use this variable $branch() to
> set branch flags to each indivisual branch depending on its index 0 for
> branch 0 and 1 for branch 1.
>
> use branch routing blocks to execute certain logic (it could be the
> manipulation ) for each branch before forwarding the request.
>
> // Binan
>
> --- On Thu, 8/30/12, Jayesh Nambiar  wrote:
>
> From: Jayesh Nambiar 
> Subject: [OpenSIPS-Users] Parallel forking and per branch manipulation
> To: "OpenSIPS users mailling list" 
> Date: Thursday, August 30, 2012, 8:11 AM
>
> Hi All,I have a scenario to do parellel forking between the user found in
> location and towards a PSTN gateway. The requirement here is, I have unique
> manipulations to be done before routing the call which I believe can be
> done using branch routes, but I'm just wondering how.?
> For eg: The branch that goes to the user found from location should have
> no FROM Header manipulation and the branch that goes out to PSTN gateway
> needs a uac_replace_from("sip:123456789@1.2.3.4"), and
> insert_hf("P-Asserted_Identity: sip:123456789@1.2.3.4\r\n") before
> calling t_relay(). There are many other manipulations like this.
> How do I take the script logic to individual branch routes before calling
> t_relay, because whats happening with me is once the location is looked up,
> I go ahead for manipulations that is supposed to be done for the gateway
> branch, but it gets applicable for the branch that goes out to the user as
> well !!
>
> Probably I am missing something on my understanding on how to call
> branch_routes while doing parallel forking.
> Thanks for any help.
> --- Jayesh
>
>
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Re: [OpenSIPS-Users] 403 reply forbidden

[yassine.ab]

Hi,
Sorry, I didn't understand
I used: ngrep -W byline -d eth1 INVITE

here sip INVITE  requests that is rejected by OpenSIPS:

U 192.168.0.123:5060 -> 192.168.0.72:4060
INVITE sip:yass...@open-ims.test SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport;branch=z9hG4bK808891501.
Route: .
From: "titi" ;tag=1817034708.
To: .
Call-ID: 136064409.
CSeq: 20 INVITE.
Contact: .
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE.
Max-Forwards: 70.
User-Agent: UCT IMS Client.
Subject: IMS Call.
P-Preferred-Identity: "titi" .
P-Preferred-Service: urn:xxx:3gpp-service.ims.icsi.mmtel.
Privacy: none.
P-Access-Network-Info: IEEE-802.11a.
Require: precondition.
Require: sec-agree.
Proxy-Require: sec-agree.
Supported: 100rel.
Content-Length:   334.
.
v=0.
o=- 0 0 IN IP4 192.168.0.123.
s=IMS Call.
c=IN IP4 192.168.0.123.
t=0 0.
m=audio 12494 RTP/AVP 0 8 101.
b=AS:64.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=curr:qos local none.
a=curr:qos remote none.
a=des:qos mandatory local sendrecv.
a=des:qos mandatory remote sendrecv.

#
U 192.168.0.72:4060 -> 192.168.0.123:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport=5060;branch=z9hG4bK808891501.
From: "titi" ;tag=1817034708.
To: .
Call-ID: 136064409.
CSeq: 20 INVITE.
Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.0.72:4060 "Noisy feedback tells:  pid=6288
req_src_ip=192.168.0.123 req_src_port=5060 in_uri=sip:yass...@open-ims.test
out_uri=sip:yass...@open-ims.test via_cnt==1".
.

#
U 192.168.0.72:4060 -> 192.168.0.123:5060
SIP/2.0 403 Preload Route denied.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport=5060;branch=z9hG4bK808891501.
From: "titi" ;tag=1817034708.
To: ;tag=d9d72f9546b6b56e4e4be7c5ecbf45ae.0cec.
Call-ID: 136064409.
CSeq: 20 INVITE.
Server: OpenSIPS (1.8.1-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.0.70:5060 "Noisy feedback tells:  pid=5915
req_src_ip=192.168.0.72 req_src_port=6060 in_uri=sip:yass...@open-ims.test
out_uri=sip:yass...@open-ims.test via_cnt==3".
.

Thanks
regards 
yassine




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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Kevin Mathy]

Hi Bogdan,

We've got some news about our problem with record-routes.

In fact, it seems that if we change Contact header sent by customer's
device, everything work fine. Explanations :

Before, in all message sent by customer's device, particularly 200OK,
Contact header was like *Contact:
*
In this situation, OpenSIPS was unable to route correctly ACK messages
following this 200OK.

Then, we've change the manner which Contact header is sent, and now it's
like *Contact: *
And in this situation, everything seems to be OK, all message, including
ACK, are correctly routed.

Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN returns
exactly CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS resolution
problem...

If it can help you !

Thanks a lot,

*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr

Pour toute demande de support, merci de contacter le *03.51.08.42.07*, ou
bien d'adresser un e-mail à *supp...@hexanet.fr*




2012/8/29 

> Hi Bogdan,
> we will do this debug before end of week or begin of next week and we will
> send our results.
>
> bye
>
> > Hi Kevin,
> >
> > This looks like OpenSIPS does not recognize the Route as its own IPs and
> > also seeing the next hop as a strict router.
> >
> > To sort this out in the fastest way, see my prev request on the logs for
> > ACK processing (with the debug=6).
> >
> > Regards,
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> >
> > On 08/28/2012 03:22 PM, Kevin Mathy wrote:
> >> Hi Bogdan,
> >>
> >> I'm working with Mickael about this problem, and we have some
> >> informations which may help you (and then help us ;-) ) :
> >>
> >> We have found that "loose_route" function modify the Request-URI
> >> variable ($ru), as you can see below :
> >>
> >> ACK message comes from provider, with $ru = sip:+333@6.6.6.6
> >> 
> >> After, loose_route function is executed, and $ru become like $ru
> >> =
> >>
> sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5
> >>
> >> The last $ru value results from a Route header
> >>
> >> For information, Record-route of previous message (200OK) is composed
> >> with two record-route in the same field, comma separated.
> >> Is Opensips 1.6.4 able to interpret this type of Record-route ?
> >>
> >> Is loose_route function using Route headers of previous messages
> >> (200OK before ACK) to route this message ? Or is it using only actual
> >> message's Route headers ?
> >>
> >> Thanks in advance,
> >> If you need further informations, feel free to ask us.
> >>
> >> Regards,
> >>
> >> *Kevin MATHY*
> >> *HEXANET*
> >> *
> >> --
> >> *
> >> Téléphone : 03.26.79.30.05
> >> Web : www.hexanet.fr 
> >>
> >> Pour toute demande de support, merci de contacter le *03.51.08.42.07*,
> >> ou bien d'adresser un e-mail à *supp...@hexanet.fr
> >> 
>
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Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[Nick Khamis]

>> RTP is used for many purposes most commonly to hide your internal network 
>> topology i.e. in your case your * server also it helps in NAT traversal.

That is exactly what we are trying to accomplish. Would like all SIP
traffic to go through the OpenSIPS server,


Could someone kindly illustrate graphically how the flow of traffic is
suppose to look in a topology hidden network using NAT traversal
please? Should there be port forwarding
of RTP traffic to the Proxy? Meaning, should the incoming ITSP traffic
be directed at OpenSIPS:


->: Initial Stream
*> Asterisk Stream

What I think we have now, and why we don't need RTP NAT traversal yet:

Scenario 1

UC > Proxy > Asterisk > ITSP > * Router 192.168.2.1  * > Back to
Asterisk > Back to Proxy > Back to UC (Final)

  Proxy--->Asterisk>  ITSP
UC  ---> *   *  *   *
  <** *  ** Router***


What we should have and need eventually (Topology Hiding using NAT traversal):

Scenario 2

UC > Proxy > Asterisk > ITSP >  *Router 192.168.2.1 * > Back to
OpenSIPS > Back to UC (Final)


  Proxy--->Asterisk>ITSP
UC  ---> * *
  <** * *
 *Router 192.168.2.1



i.e., there has to be no direct connection between the UCs or ITSP and
asterisk. I think we have accomplished this without using RTP proxy
since everything is on the same subnet,
or am I incorrect? And as you mentioned, the RTP stream is now being
directed at the Asterisk server? Will get a NGREP trace posted when I
get to work. Please remember
that this is done using virtual machines as a test bed for now...



Thank you Kindly for your Help,

Nick.


On Fri, Aug 31, 2012 at 12:11 AM, qasimak...@gmail.com
 wrote:
> Most probably your RTP stream is directly bein connected to * server. RTP is
> used for many purposes most commonly to hide your internal network topology
> i.e. in your case your * server also it helps in NAT traversal.
>
> Regards,
> Qasim
>
> On Fri, Aug 31, 2012 at 8:05 AM, Nick Khamis  wrote:
>>
>> Thank you guys for your response. It's the strangest thing. I removed
>> all the rtpproxy and nathelper stuff, and it works perfectly?
>> Everything is on the same subnet, i.e.:
>>
>> Router
>> |
>> - MySQL 192.168.2.105 (Virtual Box)
>> |
>> - OpenSIPS 192.168.2.102 (Virtual Box)
>> |
>> -Asterisk 192.168.2.110 (Virtual Box)
>> |
>> - Polycom 192.168.2.102 (Virtual Box)
>>
>>
>> The flow of transmition is:
>>
>> UC--->Proxy--->Asterisk
>>
>>
>> And I have two way audio without rtp proxy or port forwarding. I'm now
>> scared and confused. Can someone please
>> explain to me how this is?
>>
>> Thanks in Advnace,
>>
>> Nick.
>>
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Re: [OpenSIPS-Users] A Dynamic List of Gateways

[Ali Pey]

My apologies Shaahin for misspelling your name. I actually know this name
very well and should have paid more attention :)

Good luck with your design, sounds like you have some exciting time ahead
of you.

Regards,
Ali Pey

On Thu, Aug 30, 2012 at 10:55 PM, Shaahin Madani
wrote:

> Greetings,
>
> Thanks for the helpful replies. Presently, the list of gateways cannot be
> retrieved using a direct query on the external database, and hence I need
> the Perl script to act as the middle-ware too. Some changes in the external
> database may fix this, and then I would be able to use the avops module
> as Ali suggested, but I need to first thoroughly study all the
> consequences ...
>
> Regarding the bottleneck issue, I don't see any way that I can avoid the
> queries (or Perl). In case it turns out to be a real bottleneck, then is
> there any way to go around it, or will I actually be hitting an OpenSIPS
> limitation?
>
> Cheers,
> Shaahin
>
> P.S. [to Ali and Rudy]: It's "Shaahin", not "Shaheen" :-)
>
>
>
> On Fri, Aug 31, 2012 at 2:03 AM, Rudy  wrote:
>
>> Brett,
>>
>>  Any complex logic, perl, sql or otherwise can impact performance and
>> create a bottleneck. Regarding the perl module in particular, I can
>> tell you that it works pretty well.
>>
>> Thanks in advance,
>> --Rudy
>> Dynamic Packet
>> Toll-Free: 888.929.VOIP ( 8647 )
>>
>>
>> On Thu, Aug 30, 2012 at 11:58 AM, Brett Nemeroff 
>> wrote:
>> > If you can sum up the logic in an SQL query I'd recommend doing that.
>> If you
>> > can cache the results in memcache, even better. Just keep in mind that
>> any
>> > complicated logic can serve as a significant bottleneck. Depending on
>> your
>> > traffic load, that may or may not be an issue.
>> >
>> > I know there are a few recommendations for the perl module here, but
>> I'm not
>> > sure of the status of the module and if it's actively maintained. I'd
>> love
>> > to hear from others about their own experiences using it. I haven't
>> used it
>> > myself in maybe 6 years or so.
>> > -Brett
>> >
>> >
>> > On Thu, Aug 30, 2012 at 9:09 AM, Shaahin Madani <
>> shaahin.mad...@gmail.com>
>> > wrote:
>> >>
>> >> Hi Bogdan and Brett,
>> >>
>> >> Thanks for your replies. The issue here is that my OpenSIPS simply does
>> >> not have the list of gateways, that is the gateways are *not*
>> available in
>> >> the dr_gateways table. For every INVITE request, the list of gateways
>> must
>> >> be dynamically built, and only afterwards the actual dynamic routing
>> can
>> >> take place. The logic behind building the list of gateways relies on a
>> >> totally separate database.
>> >> To sum it up, I would say the desired scenario would resemble the steps
>> >> below:
>> >>
>> >> 1) OpenSIPS receives an INVITE
>> >> 2) OpenSIPS asks an external system (e.g. the Perl script) for the
>> >> gateways available for this specific INVITE request
>> >> 3) A list of gateways is returned to OpenSIPS (based on some black-box
>> >> logic)
>> >> 4) OpenSIPS dynamically routes the INVITE to the specified gateways,
>> >> prioritising based on cost, time, or ...
>> >>
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
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Re: [OpenSIPS-Users] 403 reply forbidden

[Bogdan-Andrei Iancu]

Hi,

So, is your opensips recognizing "scscf.open-ims.test:6060" as a local 
domain ? A valid preloaded route has to point to the first hop (which in 
this case is your opensips).


You can try using the "alias" statement in your cfg to map 
"scscf.open-ims.test:6060" over your opensips:


alias= scscf.open-ims.test:6060

see http://www.opensips.org/Resources/DocsCoreFcn17#toc26

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/31/2012 03:27 PM, yassine.ab wrote:

Hi,
Sorry, I didn't understand
I used: ngrep -W byline -d eth1 INVITE

here sip INVITE  requests that is rejected by OpenSIPS:

U 192.168.0.123:5060 ->  192.168.0.72:4060
INVITE sip:yass...@open-ims.test SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport;branch=z9hG4bK808891501.
Route:.
From: "titi";tag=1817034708.
To:.
Call-ID: 136064409.
CSeq: 20 INVITE.
Contact:.
Content-Type: application/sdp.
Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE.
Max-Forwards: 70.
User-Agent: UCT IMS Client.
Subject: IMS Call.
P-Preferred-Identity: "titi".
P-Preferred-Service: urn:xxx:3gpp-service.ims.icsi.mmtel.
Privacy: none.
P-Access-Network-Info: IEEE-802.11a.
Require: precondition.
Require: sec-agree.
Proxy-Require: sec-agree.
Supported: 100rel.
Content-Length:   334.
.
v=0.
o=- 0 0 IN IP4 192.168.0.123.
s=IMS Call.
c=IN IP4 192.168.0.123.
t=0 0.
m=audio 12494 RTP/AVP 0 8 101.
b=AS:64.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=curr:qos local none.
a=curr:qos remote none.
a=des:qos mandatory local sendrecv.
a=des:qos mandatory remote sendrecv.

#
U 192.168.0.72:4060 ->  192.168.0.123:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport=5060;branch=z9hG4bK808891501.
From: "titi";tag=1817034708.
To:.
Call-ID: 136064409.
CSeq: 20 INVITE.
Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.0.72:4060 "Noisy feedback tells:  pid=6288
req_src_ip=192.168.0.123 req_src_port=5060 in_uri=sip:yass...@open-ims.test
out_uri=sip:yass...@open-ims.test via_cnt==1".
.

#
U 192.168.0.72:4060 ->  192.168.0.123:5060
SIP/2.0 403 Preload Route denied.
Via: SIP/2.0/UDP 192.168.0.123:5060;rport=5060;branch=z9hG4bK808891501.
From: "titi";tag=1817034708.
To:;tag=d9d72f9546b6b56e4e4be7c5ecbf45ae.0cec.
Call-ID: 136064409.
CSeq: 20 INVITE.
Server: OpenSIPS (1.8.1-notls (i386/linux)).
Content-Length: 0.
Warning: 392 192.168.0.70:5060 "Noisy feedback tells:  pid=5915
req_src_ip=192.168.0.72 req_src_port=6060 in_uri=sip:yass...@open-ims.test
out_uri=sip:yass...@open-ims.test via_cnt==3".
.

Thanks
regards
yassine




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Re: [OpenSIPS-Users] External SIP require AUTH

[Ivan Milivojevic]

http://www.opensips.org/html/docs/modules/devel/uac_registrant.html


-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of spady
Sent: Tuesday, August 28, 2012 3:47 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] External SIP require AUTH

Hi all, maybe is a dummy question but I don't know how to achieve solution.
I am connecting a External SIP LINE provided by a Pubblic ISP.
It require authentication for each incoming call ( outgoing call by opensips
point of view ).
How can i solve it? Is there a module or a db table where store credentials
and then use them by a snippet of code??
Help,
thanks




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[OpenSIPS-Users] SIP ALG Detector

[Brett Hunter]

This tool could be a necessary part of my everyday job, if I could get it to 
work! Can anyone provide assistance with using this tool? I continue to receive 
the following for the client:
[cid:image001.png@01CD84FB.440BE3A0]

And I continually receive the following for the server:
[cid:image002.png@01CD84FB.B0B852E0]

I do not know a lot about Ruby, but this appears to be failing either because I 
do not have all the Ruby components or do not have Ruby installed correctly. 
Any help would be greatly appreciated!

Regards,
Brett Hunter
Network Engineer

Phone: (563) 445-4270
Fax: (563) 445-4271
Service: (563) 326-1237

[cid:image003.jpg@01CD84FB.E1313A90]
"We Can Do That!"
http://www.midlandcom.com/




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Re: [OpenSIPS-Users] disabling GW in gawteways table

[Joshua Laroff]

Yes if you disable a gateway in the dr_gateways table then that
gateway will be disabled if it is included in the  carrier in
dr_carriers.
It's easy to test this. Give it a whirl.
Josh

On Wed, Aug 29, 2012 at 6:35 AM, Engineer voip  wrote:
> Hello All,
>
> if i disable the GW1 in Gateways table but i use it in dr_carriers
> (dr_gw_list) table
>
> and i want to know if  the GW1 still receives calls from dr_carriers table.
> ??
>
> thank you
>
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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Bogdan-Andrei Iancu]

Hi Kevin,

This looks like OpenSIPS does not recognize the Route as its own IPs and 
also seeing the next hop as a strict router.


To sort this out in the fastest way, see my prev request on the logs for 
ACK processing (with the debug=6).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/28/2012 03:22 PM, Kevin Mathy wrote:

Hi Bogdan,

I'm working with Mickael about this problem, and we have some 
informations which may help you (and then help us ;-) ) :


We have found that "loose_route" function modify the Request-URI 
variable ($ru), as you can see below :


ACK message comes from provider, with $ru = sip:+333@6.6.6.6 

After, loose_route function is executed, and $ru become like $ru 
= sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5


The last $ru value results from a Route header

For information, Record-route of previous message (200OK) is composed 
with two record-route in the same field, comma separated.

Is Opensips 1.6.4 able to interpret this type of Record-route ?

Is loose_route function using Route headers of previous messages 
(200OK before ACK) to route this message ? Or is it using only actual 
message's Route headers ?


Thanks in advance,
If you need further informations, feel free to ask us.

Regards,

*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr 

Pour toute demande de support, merci de contacter le *03.51.08.42.07*, 
ou bien d'adresser un e-mail à *supp...@hexanet.fr 
*





2012/8/27 Bogdan-Andrei Iancu >


As the next destination should be 6.6.6.6, some stupid questions:

1) is 6.6.6.6 configured in domain module or as "alias" ?

2) have you check to loopback interface ? maybe the ACK is
spiraling over there.

Also, try to reproduce the ACK case, but enable full debug in
script for the loose_route() function:
if (is_method=="ACK") setdebug(6);

And post the output of opensips - this will give some clue on what
it is doing.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/27/2012 06:41 PM, mick...@winlux.fr
 wrote:

7.7.7.7 and 8.8.8.8 are both from the same instance of
opensips (same
server 2 IPs interfaces).

in opensips log, no error, just no forwarding to customer IPBX.
but tomorrow I'll test with other debug option.

Thanks

The 7.7.7.7 and 8.8.8.8 IPs are both from the same
instance of opensips
(doing interface exchange) or there are 2 different SIP
entities ?

Also, do you see any errors in the logs when the ACK is
handled ?

regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/27/2012 05:58 PM, mick...@winlux.fr
 wrote:

Hi
in attachement call flow and ACK from provider (this
ACK is not resend
by
opensips to customer IPBX).

Thanks

Hi,

Please post the ACK message your opensips proxy
gets, and also the
outbound ACK request (leaving your opensips) -
Include also the net
layer info (like src ip and port) - feel free to
masquerade the IPs,
but
do it consistently,

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/27/2012 09:17 AM, mick...@winlux.fr
 wrote:

Hi,
do you think this may be the cause of my
routing ACK problem ?

Hi,
loose route parameter lr , which can be
present in sip or sips
Record-Route and Route URIs to indicate
that the proxy server
identified
by the URI supports loose routing.


RFC 3261 explains the "lr" parameter as
just ";lr", not lr=on. This
brokes
some UAs which add =on to the "lr".
 opensips adds=on  to be
compatible
with these UAs


//Binan.


   

Re: [OpenSIPS-Users] Lookup location problem on OpenSIPS(multi-domain) + Asterisks

[Bogdan-Andrei Iancu]

Hi Kook,

If you want to use multiple domains, then the domain info must be 
consistent between registration and lookup - I mean users should 
register and be searched inside the same domain (for each user).


What domain do users A and B use for registering into OpenSIPS ? Also 
what is the domain in RURI when looking up for user B in OpenSIPS (for 
the call received from Asterisk) ?


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/29/2012 12:05 PM, Kook Maeng wrote:

Hi, all.

My goal is to building scalable multi-domain(tenant) telephone system 
using Opensips with multiple asterisk servers that can be multi-domain.

But I ran into a problem during the progress.

Structure of the system for test is
Opensips(10.10.10.10) -- Asterisk A(20.02.0.20), Asterisk 
B(30.30.30.30)


My Opensips modules(auth_db, usrloc, uri) are enabled "use_domain".
Below are flows for test.

Flow 1. Registrar
a : User A (1...@foo.com ) -register-> 
OpenSIPS  => 200 OK.
b : User B (2...@bar.com ) -register-> 
OpenSIPS  => 200 OK.


Flow 2. User A(1...@foo.com ) call User 
B(2...@bar.com )
a :  User A -INVITE-> OpenSIPS (via load_balancer) 
-INVITE-> Asterisk A
b :  Asterisk A -Dial(SIP/${EXTEN}@OPENSIPS)-> OpenSIPS => 
lookup("location) FAILED about 2000@10.10.10.10 


If "usrloc.so" dont't use domain it works good. but otherwise it fails.
I think the problem occurred because the lookup() find contact using 
"userid@domain".


How can I fix this? I will append content of opensips.cfg below.

Would really appreciate it if you can help.
Thanks in advance.

-Kook Maeng


- opensip.cfg -

### Global Parameters #

alias="14.63.222.202"
advertised_address="14.63.222.202"

#debug=3
#log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

port=5060
#listen=udp:211.62.253.136


### Modules Section 

#set module path
mpath="/usr/lib/opensips/modules/"

loadmodule "db_mysql.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri.so"
loadmodule "acc.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "alias_db.so"
loadmodule "domain.so"
loadmodule "permissions.so"
loadmodule "dialog.so"
loadmodule "load_balancer.so"
loadmodule "nathelper.so"
loadmodule "sipmsgops.so"
loadmodule "group.so"

# - setting module-specific parameters ---

# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")

# - rr params -
modparam("rr", "append_fromtag", 0)

# - registrar params -
modparam("registrar", "max_expires", 60)
modparam("registrar", "min_expires", 60)

# - usrloc params -
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("usrloc", "nat_bflag", 6)
modparam("usrloc", "use_domain", 1)
modparam("usrloc", "timer_interval", 30)

# - uri params -
modparam("uri", "use_uri_table", 0)
modparam("uri", "use_domain", 1)

# - acc params -
modparam("acc", "early_media", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)

# - auth_db params -
modparam("auth_db", "calculate_ha1", 0)
#modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", 1)

# - alias_db params -
modparam("alias_db", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")

# - domain params -
modparam("domain", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("domain", "db_mode", 1)   # Use caching

# - multi-module params -

# - dialog params -
modparam("dialog", "db_url", 
"mysql://opensips:opensipsrw@localhost/opensips")

modparam("dialog", "db_mode", 1)
#modparam("dialog", "dlg_flag", 4)

# - load_balancer params -
modparam("load_balancer", "db_url", 
"mysql://opensips:opensipsrw@localhost/opensips")


# - nathelper params -
modparam("registrar","received_avp", "$avp(i:42)")
modparam("nathelper","received_avp", "$avp(i:42)")

# - permissions params -
modparam("permissions", "db_url", 
"mysql://opensips:opensipsrw@localhost/opensips")


# - group params -
modparam("group", "db_url", 
"mysql://opensips:opensipsrw@localhost/opensips")

modparam("group", "use_domain", 1)



### Routing Logic 


# main request routing logic

Re: [OpenSIPS-Users] Can't login the control panel due to insufficient permissions

[microx]

Dear Ali, 

Thanks for your reply. However, I am not sure about what you mean. From the
installation guide, I see nothing 
about the modification to the local.inc.php file. Would you please be more
specific? Appreciate your help deeply.


Best regards,
Chen-Che Huang



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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Binan AL Halabi]

Hi Bogdan,

i see in the opensips Docs the definition of loose_route() function: 

" The function performs routing of SIP requests which contain a route 
set. The name is a little bit confusing, as this function also 
routes 
requests which are in the “strict router” format.
"

so Does this function do the strict routing also ? according to which chapter 
in RFC this routing belong ?

regards

--- On Tue, 8/28/12, Bogdan-Andrei Iancu  wrote:

From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] [Re:  Routing problem with Record-Route]
To: "OpenSIPS users mailling list" 
Cc: "Binan AL Halabi" 
Date: Tuesday, August 28, 2012, 7:29 AM



  


  
  
Hi,

  

  That is not true - OpenSIPS acts all the time as a loose router.
  The param you mentioned simply changes on how the "loose router"
  label should be advertised : (1) "lr" or (2) "lr=on" - but both do
  mean a loose router.

  

  Regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/28/2012 03:46 PM, Binan AL Halabi wrote:

  

  
Hi , 

  the statment *modparam("rr", "enable_full_lr", 1)*  in the
  script lets opensips uses lr=on instead of just ;lr to
  work as loose router, so it behaves as strict router where
  it should be loose router   in fail case, since it detects
  only ;lr in messages.   

  

  

  --- On Mon, 8/27/12, Bogdan-Andrei Iancu 
  wrote:

  

From: Bogdan-Andrei Iancu 

Subject: Re: [OpenSIPS-Users] [Re: Routing problem with
Record-Route]

To: mick...@winlux.fr

Cc: "OpenSIPS users mailling list"


Date: Monday, August 27, 2012, 8:48 AM



As the next destination should be
  6.6.6.6, some stupid questions:

  

  1) is 6.6.6.6 configured in domain module or as
  "alias" ?

  

  2) have you check to loopback interface ? maybe the
  ACK is spiraling 

  over there.

  

  Also, try to reproduce the ACK case, but enable full
  debug in script for 

  the loose_route() function:

       if (is_method=="ACK") setdebug(6);

  

  And post the output of opensips - this will give some
  clue on what it is 

  doing.

  

  Regards,

  

  Bogdan-Andrei Iancu

  OpenSIPS Founder and Developer

  http://www.opensips-solutions.com

  

  

  On 08/27/2012 06:41 PM, mick...@winlux.fr
  wrote:

  > 7.7.7.7 and 8.8.8.8 are both from the same
  instance of opensips (same

  > server 2 IPs interfaces).

  >

  > in opensips log, no error, just no forwarding to
  customer IPBX.

  > but tomorrow I'll test with other debug option.

  >

  > Thanks

  >

  >> The 7.7.7.7 and 8.8.8.8 IPs are both from the
  same instance of opensips

  >> (doing interface exchange) or there are 2
  different SIP entities ?

  >>

  >> Also, do you see any errors in the logs when
  the ACK is handled ?

  >>

  >> regards,

  >>

  >> Bogdan-Andrei Iancu

  >> OpenSIPS Founder and Developer

  >> http://www.opensips-solutions.com

  >>

  >>

  >> On 08/27/2012 05:58 PM, mick...@winlux.fr
  wrote:

  >>> Hi

  >>> in attachement call flow and ACK from
  provider (this ACK is not resend

  >>> by

  >>> opensips to customer IPBX).

  >>>

  >>> Thanks

  >>>

   Hi,

  

   Please post the ACK message your
  opensips proxy gets, and also the

   outbound ACK request (leaving your
  opensips) - Include also the net

   layer info (like src ip and port) -
  feel free to masquerade the IPs,

   but

   do it consistently,

  

  

Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Bogdan-Andrei Iancu]

Hi,

OpenSIPS all the time adds is Route as a loose route, so for its own 
Route, when doing loose_route() will act as loose router all the time. 
Nevertheless, when checking the next hop, OpenSIPS can handle both a 
strict or loose router (as next hop).


But once again, opensips itself does only loose routing for its Routes.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/28/2012 05:48 PM, Binan AL Halabi wrote:

Hi Bogdan,

i see in the opensips Docs the definition of loose_route() function:

" The function performs routing of SIP requests which contain a route 
set. The name is a little bit confusing, as *this function also routes 
requests which are in the “strict router” format. "*


so Does this function do the strict routing also ? according to which 
chapter in RFC this routing belong ?


regards

--- On *Tue, 8/28/12, Bogdan-Andrei Iancu //* wrote:


From: Bogdan-Andrei Iancu 
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "OpenSIPS users mailling list" 
Cc: "Binan AL Halabi" 
Date: Tuesday, August 28, 2012, 7:29 AM

Hi,

That is not true - OpenSIPS acts all the time as a loose router.
The param you mentioned simply changes on how the "loose router"
label should be advertised : (1) "lr" or (2) "lr=on" - but both do
mean a loose router.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/28/2012 03:46 PM, Binan AL Halabi wrote:

Hi ,
the statment *modparam("rr", "enable_full_lr", 1)*  in the script
lets opensips uses lr=on instead of just ;lr to work as loose
router, so it behaves as strict router where it should be loose
router   in fail case, since it detects only ;lr in messages.


--- On *Mon, 8/27/12, Bogdan-Andrei Iancu /
/* wrote:


From: Bogdan-Andrei Iancu 

Subject: Re: [OpenSIPS-Users] [Re: Routing problem with
Record-Route]
To: mick...@winlux.fr 
Cc: "OpenSIPS users mailling list" 

Date: Monday, August 27, 2012, 8:48 AM

As the next destination should be 6.6.6.6, some stupid questions:

1) is 6.6.6.6 configured in domain module or as "alias" ?

2) have you check to loopback interface ? maybe the ACK is
spiraling
over there.

Also, try to reproduce the ACK case, but enable full debug in
script for
the loose_route() function:
 if (is_method=="ACK") setdebug(6);

And post the output of opensips - this will give some clue on
what it is
doing.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/27/2012 06:41 PM, mick...@winlux.fr wrote:
> 7.7.7.7 and 8.8.8.8 are both from the same instance of
opensips (same
> server 2 IPs interfaces).
>
> in opensips log, no error, just no forwarding to customer IPBX.
> but tomorrow I'll test with other debug option.
>
> Thanks
>
>> The 7.7.7.7 and 8.8.8.8 IPs are both from the same
instance of opensips
>> (doing interface exchange) or there are 2 different SIP
entities ?
>>
>> Also, do you see any errors in the logs when the ACK is
handled ?
>>
>> regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 08/27/2012 05:58 PM, mick...@winlux.fr wrote:
>>> Hi
>>> in attachement call flow and ACK from provider (this ACK
is not resend
>>> by
>>> opensips to customer IPBX).
>>>
>>> Thanks
>>>
 Hi,

 Please post the ACK message your opensips proxy gets,
and also the
 outbound ACK request (leaving your opensips) - Include
also the net
 layer info (like src ip and port) - feel free to
masquerade the IPs,
 but
 do it consistently,

 Regards,

 Bogdan-Andrei Iancu
 OpenSIPS Founder and Developer
 http://www.opensips-solutions.com


 On 08/27/2012 09:17 AM, mick...@winlux.fr wrote:
> Hi,
> do you think this may be the cause of my routing ACK
problem ?
>
>> Hi,
>> loose route parameter lr , which can be present in sip
or sips
>> Record-Route and Route URIs to indicate that the proxy
server
>> identified
>> by the URI supports loose routing.
>>
>>
>> RFC 3261 explains the "lr" parameter as just ";lr",
not lr

Re: [OpenSIPS-Users] OpenSIPS Crashes

[Jorge Ortea]

Hi Bogdan,

I have been investigated it, I got the ERROR logs lines, I don't know if this 
can help you: 


Aug 28 01:57:03 muc-vfk21 /usr/local/opensips/sbin/opensips[3775]: 
ERROR:core:tcp_send: connect failed 
Aug 28 01:57:03 muc-vfk21 /usr/local/opensips/sbin/opensips[3775]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 01:57:03 muc-vfk21 /usr/local/opensips/sbin/opensips[3775]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_send: connect failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3805]: 
ERROR:core:_tls_read: SYSCALL error -> (024)  
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3805]: 
ERROR:core:_tls_read: something wrong in SSL: 5 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3805]: 
ERROR:core:tcp_read_req: failed to read  
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_send: connect failed 
Aug 28 01:57:50 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_send: connect failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:core:tcp_send: connect failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 01:57:51 muc-vfk21 /usr/local/opensips/sbin/opensips[3762]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_send: connect failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:tm:msg_send: tcp_send failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:tm:t_forward_nonack: sending request failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_blocking_connect: poll error: flags 18 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection 
refused 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcpconn_connect: tcp_blocking_connect failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin/opensips[3764]: 
ERROR:core:tcp_send: connect failed 
Aug 28 02:00:05 muc-vfk21 /usr/local/opensips/sbin

[OpenSIPS-Users] Opensips as front end for late offer handling

[sindrek]

Hi,

I'm working for a firm which are using the sip porta platform. This is fine
despite it does not support late offer. That is a client/pbx etc sending an
initial invite and possible re-invites with no sdp. This is legal, but
portaone does not support it (they will in a year or so, but until then ..).

The idea then is to put some sort of front end for these types of endpoints
and as a man in the middle ... detect invites with no sdp and insert a fake
sdp into it before it continues to the porta platform. This part is ok and I
have made it and it works.

The problem is all the other signalling, it sounds simple, but it is not
since all the signalling will go through this front end. The clients/pbxs
wil be directed to register and signal everything towards the front end's ip
address.

I have tried to do this for some weeks now using opensips as a proxy (not
b2bua). What I did was to let the register msg make entries in the front end
opensips proxy and then fwd the register message to porta which does all the
auth etc. 

If the registration is ok I'll keep it in the location table in opensips. If
not I wipe it again. This works fine, but what is difficult further is to
handle all the other signalling with replies and everything which passes
through. 

Porta thinks all endpoints that are registered via the opensips front end is
at the opensips front end's ip (the only way). Opensips them needs a lot of
logic to rewrite this and that from what it got in the location table (real
endpoint ip etc).  I came a step forward, but it gets more and more
complicated and I have concluded to stop for now.

Right now I'm looking into using opensips as a transparent b2bua instead
just looking for invites with no sdp. I'm not sure if this is possible or
simpler yet since I'm new to configuring it as a b2bua. So I'm asking you
experts for some advice .. ? 

Is this possible using opensips you think? If so in b2bua mode? If so again
any b2bua config examples would be very interesting to me :-) 

/BR
Sindre



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Re: [OpenSIPS-Users] textops replace and variables ..

[sindrek]

Hi thanks a lot!

I solved it :-)

/BR
Sindre



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[OpenSIPS-Users] [OpenSIPS Security Alerts] [FIX] [Severity Medium] Dialog module

[Vlad Paiu]

This message was generated by the Security Alerts service ( Free Trial 14th of 
August - 14th of September )
http://www.opensips.org/Resources/AlertsMain
*
SVN commit*:
http://opensips.svn.sourceforge.net/viewvc/opensips?view=revision&revision=9197

*Severity*: Medium

*Version*  : 1.8, trunk

*Affected modules*  : Dialog

*Effect*  : Distributed Load-Balacing was not working properly

*Affected scenarios*: In a distributed load-balancing setup, the LoadBalancer 
module
was not properly marking the internally-created dialog profiles as shared,
thus breaking the distributed functionality.

*Description:*  The bug was in the Dialog module. When the LoadBalancer modules 
used the
Dialog API to create distributed profiles, the Dialog module was mistakenly 
modifying
the resource name, removing the distributed marker ( /s ). When the LoadBalancer
module would search again for the profile, it couldn't be found anymore.

*Risks*  : When setting up a distributed LoadBalancer setup, each node would 
not be able
to see the other nodes load, and would end up overloading destination servers.

*Update*  :
- if you have an SVN checkout, 1.8 and trunk were fixed; so
update to a revision later than 9196 (trunk) or 9197 (1.8 branch)
- if you have OpenSIPS from sources, download and apply the patch from
http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.8/modules/dialog/dlg_profile.c?view=patch&r1=9197&r2=9196&pathrev=9197
or see the attached patch;
- if using tarballs, they were already regenerated (and include the fix)
- If using the official Debian package (apt.opensips.org), they are also
re-generated including the fix.

--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

--- modules/dialog/dlg_profile.c	2012/08/09 16:25:25	9196
+++ modules/dialog/dlg_profile.c	2012/08/09 16:27:33	9197
@@ -374,17 +374,31 @@
 struct dlg_profile_table* search_dlg_profile(str *name)
 {
 	struct dlg_profile_table *profile;
+	char *p,*e;
+	int use_cached=0;
+	str profile_name = *name;
+
+	/* check if this is a shared profile, and remove /s for lookup */
+	p = memchr(profile_name.s, '/', profile_name.len);
+
+	if (p) {
+		e = profile_name.s + profile_name.len;
+		profile_name.len = p - profile_name.s;
+		trim_spaces_lr( profile_name );
+		/* skip spaces after p */
+		for (++p; *p == ' ' && p < e; p++);
+		if ( p < e && *p == 's')
+		use_cached=1;
+	}
 
 	for( profile=profiles ; profile ; profile=profile->next ) {
-		if (name->len==profile->name.len &&
-		memcmp(name->s,profile->name.s,name->len)==0 )
+		if (profile->use_cached == use_cached && profile_name.len ==profile->name.len &&
+		memcmp(profile_name.s,profile->name.s,profile_name.len)==0 )
 			return profile;
 	}
 	return NULL;
 }
 
-
-
 static struct dlg_profile_table* new_dlg_profile( str *name, unsigned int size,
 		unsigned int has_value, unsigned use_cached)
 {

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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Binan AL Halabi]

Is CUSTOMER_DEVICE_SIP_DOMAIN  defined in opensips?

//Binan
--- On Fri, 8/31/12, Kevin Mathy  wrote:

From: Kevin Mathy 
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "Bogdan-Andrei Iancu" , "OpenSIPS users mailling list" 

Date: Friday, August 31, 2012, 5:53 AM

Hi Bogdan, 


We've got some news about our problem with record-routes.

 In fact, it seems that if we change Contact header sent by customer's device, 
everything work fine. Explanations :




Before, in all message sent by customer's device, particularly 200OK, Contact 
header was like Contact: 

In this situation, OpenSIPS was unable to route correctly ACK messages 
following this 200OK.


Then, we've change the manner which Contact header is sent, and now it's 
like Contact: 

And in this situation, everything seems to be OK, all message, including ACK, 
are correctly routed.


Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN returns exactly 
CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS resolution problem...




If it can help you !


Thanks a lot, 
Kevin MATHYHEXANET--Téléphone : 03.26.79.30.05Web : www.hexanet.fr


Pour toute demande de support, merci de contacter le 03.51.08.42.07, ou bien 
d'adresser un e-mail à supp...@hexanet.fr






2012/8/29  


Hi Bogdan,

we will do this debug before end of week or begin of next week and we will

send our results.



bye



> Hi Kevin,

>

> This looks like OpenSIPS does not recognize the Route as its own IPs and

> also seeing the next hop as a strict router.

>

> To sort this out in the fastest way, see my prev request on the logs for

> ACK processing (with the debug=6).

>

> Regards,

>

> Bogdan-Andrei Iancu

> OpenSIPS Founder and Developer

> http://www.opensips-solutions.com

>

>

> On 08/28/2012 03:22 PM, Kevin Mathy wrote:

>> Hi Bogdan,

>>

>> I'm working with Mickael about this problem, and we have some

>> informations which may help you (and then help us ;-) ) :

>>

>> We have found that "loose_route" function modify the Request-URI

>> variable ($ru), as you can see below :

>>

>> ACK message comes from provider, with $ru = sip:+333@6.6.6.6

>> 

>> After, loose_route function is executed, and $ru become like $ru

>> =

>> sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5

>>

>> The last $ru value results from a Route header

>>

>> For information, Record-route of previous message (200OK) is composed

>> with two record-route in the same field, comma separated.

>> Is Opensips 1.6.4 able to interpret this type of Record-route ?

>>

>> Is loose_route function using Route headers of previous messages

>> (200OK before ACK) to route this message ? Or is it using only actual

>> message's Route headers ?

>>

>> Thanks in advance,

>> If you need further informations, feel free to ask us.

>>

>> Regards,

>>

>> *Kevin MATHY*

>> *HEXANET*

>> *

>> --

>> *

>> Téléphone : 03.26.79.30.05

>> Web : www.hexanet.fr 

>>

>> Pour toute demande de support, merci de contacter le *03.51.08.42.07*,

>> ou bien d'adresser un e-mail à *supp...@hexanet.fr

>> 


-Inline Attachment Follows-

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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Bogdan-Andrei Iancu]

Funny...actually the problem was not that you had a FQDN in domain 
contact, but because that FQDN was considered by opensips as a local domain.


So, when openspis received the ACK with that contact in RURI, opensips 
thought is it his RR header and was acting as prev hop being a strict 
routerand this was messing up the entire routing.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/31/2012 03:53 PM, Kevin Mathy wrote:

Hi Bogdan,

We've got some news about our problem with record-routes.
In fact, it seems that if we change Contact header sent by customer's 
device, everything work fine. Explanations :


Before, in all message sent by customer's device, particularly 200OK, 
Contact header was like *Contact: 
*
In this situation, OpenSIPS was unable to route correctly ACK messages 
following this 200OK.


Then, we've change the manner which Contact header is sent, and now 
it's like *Contact: *
And in this situation, everything seems to be OK, all message, 
including ACK, are correctly routed.


Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN 
returns exactly CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS 
resolution problem...


If it can help you !

Thanks a lot,

*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr 

Pour toute demande de support, merci de contacter le *03.51.08.42.07*, 
ou bien d'adresser un e-mail à *supp...@hexanet.fr 
*





2012/8/29 mailto:mick...@winlux.fr>>

Hi Bogdan,
we will do this debug before end of week or begin of next week and
we will
send our results.

bye

> Hi Kevin,
>
> This looks like OpenSIPS does not recognize the Route as its own
IPs and
> also seeing the next hop as a strict router.
>
> To sort this out in the fastest way, see my prev request on the
logs for
> ACK processing (with the debug=6).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 08/28/2012 03:22 PM, Kevin Mathy wrote:
>> Hi Bogdan,
>>
>> I'm working with Mickael about this problem, and we have some
>> informations which may help you (and then help us ;-) ) :
>>
>> We have found that "loose_route" function modify the Request-URI
>> variable ($ru), as you can see below :
>>
>> ACK message comes from provider, with $ru =
sip:+333@6.6.6.6 
>> >
>> After, loose_route function is executed, and $ru become like $ru
>> =
>>

sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5
>>
>> The last $ru value results from a Route header
>>
>> For information, Record-route of previous message (200OK) is
composed
>> with two record-route in the same field, comma separated.
>> Is Opensips 1.6.4 able to interpret this type of Record-route ?
>>
>> Is loose_route function using Route headers of previous messages
>> (200OK before ACK) to route this message ? Or is it using only
actual
>> message's Route headers ?
>>
>> Thanks in advance,
>> If you need further informations, feel free to ask us.
>>
>> Regards,
>>
>> *Kevin MATHY*
>> *HEXANET*
>> *
>> --
>> *
>> Téléphone : 03.26.79.30.05
>> Web : www.hexanet.fr 

>>
>> Pour toute demande de support, merci de contacter le
*03.51.08.42.07*,
>> ou bien d'adresser un e-mail à *supp...@hexanet.fr

>> >

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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Binan AL Halabi]

Hi,
Is CUSTOMER_DEVICE_SIP_DOMAIN defined in opensips ?

//Binan

--- On Fri, 8/31/12, Kevin Mathy  wrote:

From: Kevin Mathy 
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "Bogdan-Andrei Iancu" , "OpenSIPS users mailling list" 

Date: Friday, August 31, 2012, 5:53 AM

Hi Bogdan, 


We've got some news about our problem with record-routes.

 In fact, it seems that if we change Contact header sent by customer's device, 
everything work fine. Explanations :




Before, in all message sent by customer's device, particularly 200OK, Contact 
header was like Contact: 

In this situation, OpenSIPS was unable to route correctly ACK messages 
following this 200OK.


Then, we've change the manner which Contact header is sent, and now it's 
like Contact: 

And in this situation, everything seems to be OK, all message, including ACK, 
are correctly routed.


Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN returns exactly 
CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS resolution problem...




If it can help you !


Thanks a lot, 
Kevin MATHYHEXANET--Téléphone : 03.26.79.30.05Web : www.hexanet.fr


Pour toute demande de support, merci de contacter le 03.51.08.42.07, ou bien 
d'adresser un e-mail à supp...@hexanet.fr






2012/8/29  


Hi Bogdan,

we will do this debug before end of week or begin of next week and we will

send our results.



bye



> Hi Kevin,

>

> This looks like OpenSIPS does not recognize the Route as its own IPs and

> also seeing the next hop as a strict router.

>

> To sort this out in the fastest way, see my prev request on the logs for

> ACK processing (with the debug=6).

>

> Regards,

>

> Bogdan-Andrei Iancu

> OpenSIPS Founder and Developer

> http://www.opensips-solutions.com

>

>

> On 08/28/2012 03:22 PM, Kevin Mathy wrote:

>> Hi Bogdan,

>>

>> I'm working with Mickael about this problem, and we have some

>> informations which may help you (and then help us ;-) ) :

>>

>> We have found that "loose_route" function modify the Request-URI

>> variable ($ru), as you can see below :

>>

>> ACK message comes from provider, with $ru = sip:+333@6.6.6.6

>> 

>> After, loose_route function is executed, and $ru become like $ru

>> =

>> sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5

>>

>> The last $ru value results from a Route header

>>

>> For information, Record-route of previous message (200OK) is composed

>> with two record-route in the same field, comma separated.

>> Is Opensips 1.6.4 able to interpret this type of Record-route ?

>>

>> Is loose_route function using Route headers of previous messages

>> (200OK before ACK) to route this message ? Or is it using only actual

>> message's Route headers ?

>>

>> Thanks in advance,

>> If you need further informations, feel free to ask us.

>>

>> Regards,

>>

>> *Kevin MATHY*

>> *HEXANET*

>> *

>> --

>> *

>> Téléphone : 03.26.79.30.05

>> Web : www.hexanet.fr 

>>

>> Pour toute demande de support, merci de contacter le *03.51.08.42.07*,

>> ou bien d'adresser un e-mail à *supp...@hexanet.fr

>> 


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Re: [OpenSIPS-Users] Error starting opensips 1.8.1

[Shimon Mishal]

Hi Odin
Could you please explain how to add modules (or load modules in Opensips)

Regards
Shimon 

From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] on 
behalf of Odin Gremaud [odin.grem...@nexcom.fr]
Sent: Thursday, August 30, 2012 6:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Error starting opensips 1.8.1

Hi,

You should add the 'sipmsgops' module to use the 'is_method' function. The
function was originally in the 'textops' module, which was split into
'textops' and 'sipmsgops'. This should solve your issue.

Hope this helps.

Regards,

Odin


On Thu, 30 Aug 2012 17:35:28 +0200, Shimon Mishal
 wrote:

>
> Hi list,
>
> I've just install the 1.8.1 version.
> But I get the following:
> Could you please advise?
>
>
> 1.  [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ opensipsctl start
>
> INFO: Starting OpenSIPS :
>
> ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
> failed
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo tail
> /var/log/messages
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 193, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 202, column 21-22: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 223, column 20-21: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 229, column 19-20: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 233, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 240, column 19-20: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 247, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 253, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 296, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: ERROR:core:main: bad config
> file (14 errors)
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$
> --
>
> 2. [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo opensipsdbctl
> create
> ERROR: database engine not specified, please setup one in the config
> script
>
> -
> 3.[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ cat
> /etc/default/opensips
> #
> # OpenSIPS startup options
> #
>
> # Set to yes to enable opensips, once configured properly.
> RUN_OPENSIPS=yes
>
> # User to run as
> USER=opensips
>
> # Group to run as
> GROUP=opensips
>
> # Amount of memory to allocate for the running OpenSIPS server (in Mb)
> MEMORY=128
>
> # Enable the server to leave a core file when it crashes.
> # Set this to 'yes' to enable OpenSIPS to leave a core file when it
> crashes
> # or 'no' to disable this feature. This option is case sensitive and only
> # accepts 'yes' and 'no' and only in lowercase letters.
> # On some systems (e.g. Ubuntu 6.10, Debian 4.0) it is necessary to
> specify
> # a directory for the core files to get a dump. Look into the opensips
> # init file for an example configuration.
> DUMP_CORE=no
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$
>
>
>
>
>
> 
> This footnote confirms that this email message has been scanned by
> PineApp Mail-SeCure for the presence of malicious code, vandals &
> computer viruses.
> 
>
>

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Re: [OpenSIPS-Users] Can't login the control panel due to insufficient permissions

[Ali Pey]

under your opensips-cp directory do this:
cd config

Then look at the files in that directory:
boxes.global.inc.php  db.inc.php  local.inc.php

db.inc.php has the db configuration and it is fairly self explanatory.

Let me know if you still have any problems.

Regards,
Ali Pey

On Wed, Aug 29, 2012 at 10:43 PM, microx  wrote:

> Dear Ali,
>
> Thanks for your reply. However, I am not sure about what you mean. From the
> installation guide, I see nothing
> about the modification to the local.inc.php file. Would you please be more
> specific? Appreciate your help deeply.
>
>
> Best regards,
> Chen-Che Huang
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Can-t-login-the-control-panel-due-to-insufficient-permissions-tp7581356p7581521.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] SIP ALG Detector

[Saúl Ibarra Corretgé]

Hi,

On Aug 28, 2012, at 4:02 PM, Brett Hunter wrote:

> This tool could be a necessary part of my everyday job, if I could get it to 
> work! Can anyone provide assistance with using this tool? I continue to 
> receive the following for the client:
> 
>  
> And I continually receive the following for the server:
> 
>  
> I do not know a lot about Ruby, but this appears to be failing either because 
> I do not have all the Ruby components or do not have Ruby installed 
> correctly. Any help would be greatly appreciated!
>  

AFAIK that tool was developed for a Unix environment so it's very unlikely that 
it runs on Windows without issues. Moreover, if you look at the error on the 
server component you'll see that the fork() syscall is apparently not 
implemented for (at least) that version of Windows.


Regards,

--
Saúl Ibarra Corretgé
AG Projects




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[OpenSIPS-Users] FW: Error starting opensips 1.8.1

[Shimon Mishal]

Hi Odin
Could you please explain how to add modules (or load modules in Opensips)
Please describe .

Regards
Shimon

From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] on 
behalf of Odin Gremaud [odin.grem...@nexcom.fr]
Sent: Thursday, August 30, 2012 6:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Error starting opensips 1.8.1

Hi,

You should add the 'sipmsgops' module to use the 'is_method' function. The
function was originally in the 'textops' module, which was split into
'textops' and 'sipmsgops'. This should solve your issue.

Hope this helps.

Regards,

Odin


On Thu, 30 Aug 2012 17:35:28 +0200, Shimon Mishal
 wrote:

>
> Hi list,
>
> I've just install the 1.8.1 version.
> But I get the following:
> Could you please advise?
>
>
> 1.  [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ opensipsctl start
>
> INFO: Starting OpenSIPS :
>
> ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
> failed
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo tail
> /var/log/messages
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 193, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 202, column 21-22: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 223, column 20-21: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 229, column 19-20: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 233, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 240, column 19-20: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 247, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 253, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
> error in config file, line 296, column 18-19: unknown command
> , missing loadmodule?
> Aug 30 14:49:48 ip-10-128-35-32 opensips: ERROR:core:main: bad config
> file (14 errors)
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$
> --
>
> 2. [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo opensipsdbctl
> create
> ERROR: database engine not specified, please setup one in the config
> script
>
> -
> 3.[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ cat
> /etc/default/opensips
> #
> # OpenSIPS startup options
> #
>
> # Set to yes to enable opensips, once configured properly.
> RUN_OPENSIPS=yes
>
> # User to run as
> USER=opensips
>
> # Group to run as
> GROUP=opensips
>
> # Amount of memory to allocate for the running OpenSIPS server (in Mb)
> MEMORY=128
>
> # Enable the server to leave a core file when it crashes.
> # Set this to 'yes' to enable OpenSIPS to leave a core file when it
> crashes
> # or 'no' to disable this feature. This option is case sensitive and only
> # accepts 'yes' and 'no' and only in lowercase letters.
> # On some systems (e.g. Ubuntu 6.10, Debian 4.0) it is necessary to
> specify
> # a directory for the core files to get a dump. Look into the opensips
> # init file for an example configuration.
> DUMP_CORE=no
> [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$
>
>
>
>
>
> 
> This footnote confirms that this email message has been scanned by
> PineApp Mail-SeCure for the presence of malicious code, vandals &
> computer viruses.
> 
>
>

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Re: [OpenSIPS-Users] Error starting opensips 1.8.1

[Odin Gremaud]

Hello,

Use the 'loadmodule' parameter. You should already find some examples in  
your configuration file ('opensips.cfg').


Be advised that you have a complete documentation for every features in  
OpenSIPS (modules, tutorials, ...) on this page :  
http://www.opensips.org/Resources/DocsCookbooks


Regards,

Odin


On Fri, 31 Aug 2012 15:51:42 +0200, Shimon Mishal  
 wrote:



Hi Odin
Could you please explain how to add modules (or load modules in Opensips)

Regards
Shimon

From: users-boun...@lists.opensips.org  
[users-boun...@lists.opensips.org] on behalf of Odin Gremaud  
[odin.grem...@nexcom.fr]

Sent: Thursday, August 30, 2012 6:43 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Error starting opensips 1.8.1

Hi,

You should add the 'sipmsgops' module to use the 'is_method' function.  
The

function was originally in the 'textops' module, which was split into
'textops' and 'sipmsgops'. This should solve your issue.

Hope this helps.

Regards,

Odin


On Thu, 30 Aug 2012 17:35:28 +0200, Shimon Mishal
 wrote:



Hi list,

I've just install the 1.8.1 version.
But I get the following:
Could you please advise?


1.  [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ opensipsctl start

INFO: Starting OpenSIPS :

ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
failed
[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo tail
/var/log/messages
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 193, column 18-19: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 202, column 21-22: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 223, column 20-21: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 229, column 19-20: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 233, column 18-19: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 240, column 19-20: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 247, column 18-19: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 253, column 18-19: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: CRITICAL:core:yyerror: parse
error in config file, line 296, column 18-19: unknown command
, missing loadmodule?
Aug 30 14:49:48 ip-10-128-35-32 opensips: ERROR:core:main: bad config
file (14 errors)
[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$
--

2. [ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ sudo opensipsdbctl
create
ERROR: database engine not specified, please setup one in the config
script

-
3.[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$ cat
/etc/default/opensips
#
# OpenSIPS startup options
#

# Set to yes to enable opensips, once configured properly.
RUN_OPENSIPS=yes

# User to run as
USER=opensips

# Group to run as
GROUP=opensips

# Amount of memory to allocate for the running OpenSIPS server (in Mb)
MEMORY=128

# Enable the server to leave a core file when it crashes.
# Set this to 'yes' to enable OpenSIPS to leave a core file when it
crashes
# or 'no' to disable this feature. This option is case sensitive and  
only

# accepts 'yes' and 'no' and only in lowercase letters.
# On some systems (e.g. Ubuntu 6.10, Debian 4.0) it is necessary to
specify
# a directory for the core files to get a dump. Look into the opensips
# init file for an example configuration.
DUMP_CORE=no
[ec2-user@ip-10-128-35-32 opensips-1.8.1-tls]$






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**

Re: [OpenSIPS-Users] FW: Error starting opensips 1.8.1

[yassine.ab]

Hi,

first you must to see here:

/usr/local/lib/opensips/modules/

if your module is not present, you must to make:

make include_modules="your_module" modules (example: make
include_modules="db_mysql" modules)
make install include_modules="your_module" modules (example: make install
include_modules="db_mysql" modules)

after in opensips.cfg

loadmodule ".so" (example: loadmodule "db_mysql.so")
modparam()

regards 
yassine



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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Hubert Mickael]

Hi,
Yes,* CUSTOMER_DEVICE_SIP_DOMAIN *is insert to "domain" table in 
opensips's database.


Le 31/08/2012 15:48, Binan AL Halabi a écrit :

Hi,
Is *CUSTOMER_DEVICE_SIP_DOMAIN defined in opensips* ?

//Binan

--- On *Fri, 8/31/12, Kevin Mathy //* wrote:


From: Kevin Mathy 
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "Bogdan-Andrei Iancu" , "OpenSIPS users
mailling list" 
Date: Friday, August 31, 2012, 5:53 AM

Hi Bogdan,

We've got some news about our problem with record-routes.
In fact, it seems that if we change Contact header sent by
customer's device, everything work fine. Explanations :

Before, in all message sent by customer's device, particularly
200OK, Contact header was like *Contact:
*
In this situation, OpenSIPS was unable to route correctly ACK
messages following this 200OK.

Then, we've change the manner which Contact header is sent, and
now it's like *Contact: *
And in this situation, everything seems to be OK, all message,
including ACK, are correctly routed.

Further, we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN
returns exactly CUSTOMER_DEVICE_IP, so, it doesn't seems to be a
DNS resolution problem...

If it can help you !

Thanks a lot,

*Kevin MATHY*
*HEXANET*
*
--
*
Téléphone : 03.26.79.30.05
Web : www.hexanet.fr 

Pour toute demande de support, merci de contacter le
*03.51.08.42.07*, ou bien d'adresser un e-mail à
*supp...@hexanet.fr *




2012/8/29 >

Hi Bogdan,
we will do this debug before end of week or begin of next week
and we will
send our results.

bye

> Hi Kevin,
>
> This looks like OpenSIPS does not recognize the Route as its
own IPs and
> also seeing the next hop as a strict router.
>
> To sort this out in the fastest way, see my prev request on
the logs for
> ACK processing (with the debug=6).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 08/28/2012 03:22 PM, Kevin Mathy wrote:
>> Hi Bogdan,
>>
>> I'm working with Mickael about this problem, and we have some
>> informations which may help you (and then help us ;-) ) :
>>
>> We have found that "loose_route" function modify the
Request-URI
>> variable ($ru), as you can see below :
>>
>> ACK message comes from provider, with $ru =
sip:+333@6.6.6.6

>> 
>> After, loose_route function is executed, and $ru become
like $ru
>> =
>>

sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5
>>
>> The last $ru value results from a Route header
>>
>> For information, Record-route of previous message (200OK)
is composed
>> with two record-route in the same field, comma separated.
>> Is Opensips 1.6.4 able to interpret this type of Record-route ?
>>
>> Is loose_route function using Route headers of previous
messages
>> (200OK before ACK) to route this message ? Or is it using
only actual
>> message's Route headers ?
>>
>> Thanks in advance,
>> If you need further informations, feel free to ask us.
>>
>> Regards,
>>
>> *Kevin MATHY*
>> *HEXANET*
>> *
>> --
>> *
>> Téléphone : 03.26.79.30.05
>> Web : www.hexanet.fr 

>>
>> Pour toute demande de support, merci de contacter le
*03.51.08.42.07*,
>> ou bien d'adresser un e-mail à *supp...@hexanet.fr

>> 


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Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[qasimak...@gmail.com]

I dunno what you are trying to explain but a simple scenario for topology
hiding would look something liek this:

UA   OSIPs/RTPP *  CARRIER
<--<>--<>-->
<<><>##>

In the above scenario your OpenSIPs and RTPProxy are both running on the
same server so both your SIP signaling and RTP stream will be relayed to
OpenSIPs/RTP box. Now your OSIPs will relay these requests to your * server
and similarly RTPProxy will relay your RTP stream to the * box. Now if you
look on UA(User Agent's) side only one node will be visible to it. To
furthur secure your * box you can remove it from public domain and assign
it a local IP. Possibilities are endless :).

Lastly i am sure i explained every thing right. Please correct me if i am
wrong anywhere.

Regards,
Qasim


On Fri, Aug 31, 2012 at 6:20 PM, Nick Khamis  wrote:

> >> RTP is used for many purposes most commonly to hide your internal
> network topology i.e. in your case your * server also it helps in NAT
> traversal.
>
> That is exactly what we are trying to accomplish. Would like all SIP
> traffic to go through the OpenSIPS server,
>
>
> Could someone kindly illustrate graphically how the flow of traffic is
> suppose to look in a topology hidden network using NAT traversal
> please? Should there be port forwarding
> of RTP traffic to the Proxy? Meaning, should the incoming ITSP traffic
> be directed at OpenSIPS:
>
>
> ->: Initial Stream
> *> Asterisk Stream
>
> What I think we have now, and why we don't need RTP NAT traversal yet:
>
> Scenario 1
>
> UC > Proxy > Asterisk > ITSP > * Router 192.168.2.1  * > Back to
> Asterisk > Back to Proxy > Back to UC (Final)
>
>   Proxy--->Asterisk>  ITSP
> UC  ---> *   *  *   *
>   <** *  ** Router***
>
>
> What we should have and need eventually (Topology Hiding using NAT
> traversal):
>
> Scenario 2
>
> UC > Proxy > Asterisk > ITSP >  *Router 192.168.2.1 * > Back to
> OpenSIPS > Back to UC (Final)
>
>
>   Proxy--->Asterisk>ITSP
> UC  ---> * *
>   <** * *
>  *Router 192.168.2.1
>
>
>
> i.e., there has to be no direct connection between the UCs or ITSP and
> asterisk. I think we have accomplished this without using RTP proxy
> since everything is on the same subnet,
> or am I incorrect? And as you mentioned, the RTP stream is now being
> directed at the Asterisk server? Will get a NGREP trace posted when I
> get to work. Please remember
> that this is done using virtual machines as a test bed for now...
>
>
>
> Thank you Kindly for your Help,
>
> Nick.
>
>
> On Fri, Aug 31, 2012 at 12:11 AM, qasimak...@gmail.com
>  wrote:
> > Most probably your RTP stream is directly bein connected to * server.
> RTP is
> > used for many purposes most commonly to hide your internal network
> topology
> > i.e. in your case your * server also it helps in NAT traversal.
> >
> > Regards,
> > Qasim
> >
> > On Fri, Aug 31, 2012 at 8:05 AM, Nick Khamis  wrote:
> >>
> >> Thank you guys for your response. It's the strangest thing. I removed
> >> all the rtpproxy and nathelper stuff, and it works perfectly?
> >> Everything is on the same subnet, i.e.:
> >>
> >> Router
> >> |
> >> - MySQL 192.168.2.105 (Virtual Box)
> >> |
> >> - OpenSIPS 192.168.2.102 (Virtual Box)
> >> |
> >> -Asterisk 192.168.2.110 (Virtual Box)
> >> |
> >> - Polycom 192.168.2.102 (Virtual Box)
> >>
> >>
> >> The flow of transmition is:
> >>
> >> UC--->Proxy--->Asterisk
> >>
> >>
> >> And I have two way audio without rtp proxy or port forwarding. I'm now
> >> scared and confused. Can someone please
> >> explain to me how this is?
> >>
> >> Thanks in Advnace,
> >>
> >> Nick.
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> > ___
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> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
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Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

[Binan AL Halabi]

Hi Bogdan,

as that FQDN is in opensips domain table or  alias, thats why the opensips 
considerd it as internal domain , Is that correct?
It should be only in the domain table of the opensips responsible for that 
customer.

thanks
//Binan

--- On Fri, 8/31/12, Hubert Mickael  wrote:

From: Hubert Mickael 
Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]
To: "OpenSIPS users mailling list" 
Cc: "Binan AL Halabi" 
Date: Friday, August 31, 2012, 12:18 PM





  


Hi,

Yes, CUSTOMER_DEVICE_SIP_DOMAIN is insert to "domain" table in
opensips's database.



Le 31/08/2012 15:48, Binan AL Halabi a écrit :

  

  
Hi,

Is CUSTOMER_DEVICE_SIP_DOMAIN defined in opensips ?



//Binan



--- On Fri, 8/31/12, Kevin Mathy 
wrote:



From: Kevin Mathy 

Subject: Re: [OpenSIPS-Users] [Re: Routing problem with Record-Route]

To: "Bogdan-Andrei Iancu" , "OpenSIPS users
mailling list" 

Date: Friday, August 31, 2012, 5:53 AM

  

  Hi
Bogdan, 
  

  
  We've
got some news about our problem with record-routes.
   
  In
fact, it seems that if we change Contact header sent by customer's
device, everything work fine. Explanations :
  

  
  Before,
in all message sent by customer's device, particularly 200OK, Contact
header was like Contact:

  In
this situation, OpenSIPS was unable to route correctly ACK messages
following this 200OK.
  

  
  Then,
we've change the manner which Contact header is sent, and now it's like Contact:

  And
in this situation, everything seems to be OK, all message, including
ACK, are correctly routed.
  

  
  Further,
we are sure that DNS resolution of CUSTOMER_SIP_DOMAIN returns exactly
CUSTOMER_DEVICE_IP, so, it doesn't seems to be a DNS resolution
problem...
  

  
  If
it can help you !
  

  
  Thanks
a lot, 
  

  
  Kevin MATHY
  HEXANET
  
  --
  
  Téléphone : 03.26.79.30.05
  Web : www.hexanet.fr
  

  
  Pour toute demande de support, merci de contacter le 03.51.08.42.07,
ou bien d'adresser un e-mail à supp...@hexanet.fr
  

  
  

  

  

  2012/8/29 

  Hi
Bogdan,

we will do this debug before end of week or begin of next week and we
will

send our results.



bye



> Hi Kevin,

>

> This looks like OpenSIPS does not recognize the Route as its own
IPs and

> also seeing the next hop as a strict router.

>

> To sort this out in the fastest way, see my prev request on the
logs for

> ACK processing (with the debug=6).

>

> Regards,

>

> Bogdan-Andrei Iancu

> OpenSIPS Founder and Developer

> http://www.opensips-solutions.com

>

>

> On 08/28/2012 03:22 PM, Kevin Mathy wrote:

>> Hi Bogdan,

>>

>> I'm working with Mickael about this problem, and we have some

>> informations which may help you (and then help us ;-) ) :

>>

>> We have found that "loose_route" function modify the
Request-URI

>> variable ($ru), as you can see below :

>>

>> ACK message comes from provider, with $ru = sip:+333@6.6.6.6


>> 

>> After, loose_route
function is executed, and $ru become like $ru

>> =

>>
sip:7.7.7.7;lr;r2=on;ftag=c97942d9-13c4-503ca77b-ef8c9eef-760f27a5;xyz=c12.18aedaa5

>>

>> The last $ru value results from a Route header

>>

>> For information, Record-route of previous message (200OK) is
composed

>> with two record-route in the same field, comma separated.

>> Is Opensips 1.6.4 able to interpret this type of Record-route ?

>>

>> Is loose_route function using Route headers of previous
messages

>> (200OK before ACK) to route this message ? Or is it using only
actual

>> message's Route headers ?

>>

>> Thanks in advance,

>> If you need further informations, feel free to ask us.

>>

>> Regards,

>>


>> *Kevin MATHY*

>> *HEXANET*

>> *

>> --

>> *

>> Téléphone : 03.26.79.30.05

>> Web : www.hexanet.fr 

>>

>> Pour toute demande de support, merci de contacter le
*03.51.08.42.07*,

>> ou bien d'adresser un e-mail à *supp...@hexanet.fr

>> 

  
  
  
  

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