Re: [OpenSIPS-Users] about imc invite

[Schumann Sebastian]

Hi

Check whether it really sends this as text in the body of the message or if 
HTML is around. I had this problem some time ago using eyebeam and the reason 
was the the very first character was not '#' but '<' from the HTML opening tag.

Sebastian


From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] On 
Behalf Of york liu [besti...@gmail.com]
Sent: Monday, November 01, 2010 10:18
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] about imc invite

hi all

i was try to use imc module yestoday.everything work fine excluding invite 
function.

no matter sending #invite sip:j...@opensips.org 
sip:chat-...@opensips.org  or  #invite 
j...@opensips.org sent to 
sip:chat-...@opensips.org
to chat-manager.

so any sugestion to me?



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Re: [OpenSIPS-Users] about imc invite

[Schumann Sebastian]

Hi

Check whether it really sends this as text in the body of the message or if 
HTML is around. I had this problem some time ago using eyebeam and the reason 
was the the very first character was not '#' but '<' from the HTML opening tag.

Sebastian


From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] On 
Behalf Of york liu [besti...@gmail.com]
Sent: Monday, November 01, 2010 10:18
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] about imc invite

hi all

i was try to use imc module yestoday.everything work fine excluding invite 
function.

no matter sending #invite sip:j...@opensips.org 
sip:chat-...@opensips.org  or  #invite 
j...@opensips.org sent to 
sip:chat-...@opensips.org
to chat-manager.

so any sugestion to me?



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Re: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration

[Schumann Sebastian]

Hi Prem

There is a good tutorial at http://www.opensips.org/Resources/DocsTutAsterisk 
It does exactly what you need I assume.

For details in writing and extending basic configuration, you can find also the 
linked documentation there.

Best regards
Sebastian 

> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Premalatha Kuppan
> Sent: Monday, 07. June 2010 11:35
> To: OpenSIPS users mailling list
> Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK Integration
> 
> Hi,
> 
> Can anyone guide me in building the Opensips and Asterisk Integration.
> 
> I want to Use OpenSIPS as SIP PROXY (i.e all transactions and dialogs should 
> be
> handled by Opensips) and Asterisk to do only IVR functionality.
> 
> I appreciate if anyone can guide me in writing a routing logic from Opensips 
> to
> Asterisk for IVR and futher call flow OPENSIPS to handle.
> 
> Thanks,
> Prem
> 


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Re: [OpenSIPS-Users] [OpenSIPS-Devel] New features in BLF and presence

[Schumann Sebastian]

Hi Inaki

> 
> Also, if I set "offline" status (other not to see me online) and I do
> a call, will then my watchers receive a NOTIFY with status "on the
> phone"?
> 

I put the same question earlier, see Anca's answer. In short, this will be as 
you described.

> Example: I am registered (but nothing published), performing a call and hang 
> up afterwards. I assume that it will trigger from offline to online but busy 
> to offline again.
>
> Is the module considering this or would this require the use of pua_usrloc to 
> have the correct behavior for clients that do not publish presence?

The module does consider this case. The buddy will indeed appear offline 
until a call is preformed, then it will apear online with status 
'Calling' and then if answered 'On the phone' . After the call ends the 
user will appear again offline.
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Re: [OpenSIPS-Users] New features in BLF and presence

[Schumann Sebastian]

Hi Anca

Nice feature, one question:

> Publish. The result is that when a buddy is in a call you will see a
> status indicating this and in the rest of the time you will see the
> presence state that the buddy set in his client.

Is this also done for buddies that did not publish any state before? How will 
this be threaded by the module?

Example: I am registered (but nothing published), performing a call and hang up 
afterwards. I assume that it will trigger from offline to online but busy to 
offline again.

Is the module considering this or would this require the use of pua_usrloc to 
have the correct behavior for clients that do not publish presence?

Thanks!

Sebastian

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Re: [OpenSIPS-Users] SIP Presence Aggregation Issue

[Schumann Sebastian]

Hi Adrian

> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
> Sent: Thursday, 01. April 2010 16:31
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] SIP Presence Aggregation Issue
> 
> 
> On Apr 1, 2010, at 3:10 PM, Schumann Sebastian wrote:
> 
> > Hi Bogdan
> >
> > Yes, in an "ideal world", this re-publishing with BUSY and ONLINE is
> > wanted, but the toggling not. Priorities would help (you'd receive
> > that but not trigger anything on the user interface as the highest
> > priority will keep its state), but we don't have them yet, so in
> > that case the decision must be made somewhere else or not at all.
> >
> 
> All we need is a good client to figure out how to use these concepts,
> right?

I think yes, at least for this particular question.

Clients that make use of rich presence when publishing and clients that 
interpret it when creating the user state.

The question is IMHO only how to interpret it and how to display the 
information to reflect the presentity state properly.

An XMPP client has the same options IMHO (not to start a flame: I mean for this 
question, not in regard of presence implementations and standards in general): 
It could for example also display a user as busy because one resource is busy 
(maybe used by the user, hence he is busy). It could also use always the state 
if the highest prioritized resource. This depends more or less on how the 
information it receives from the service is processed and displayed. 

It does not necessarily matter if each half minutes comes a new state (BUSY and 
ONLINE by turns). As long as a) either one resource BUSY marks the whole 
presentity busy or b) the resources have priorities one consistent state would 
be shown. 

Only for the "last published" model it would alternate, but interpreting the 
last state as the current user state does not seem right for me. The last state 
_actively changed_ by the user yes, but not the last one due to refreshing a 
PUBLISH expiration.

Sebastian



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Re: [OpenSIPS-Users] SIP Presence Aggregation Issue

[Schumann Sebastian]

Hi Bogdan

Yes, in an "ideal world", this re-publishing with BUSY and ONLINE is wanted, 
but the toggling not. Priorities would help (you'd receive that but not trigger 
anything on the user interface as the highest priority will keep its state), 
but we don't have them yet, so in that case the decision must be made somewhere 
else or not at all.

In reply to Saul
>  - Try to 'fix' this in the server side: find a way to detect which is 
>the 'active' contact, that is, which is the one that is doing something 
>and only take that contact's presence information as valid. The way of 
>selecting which is the active contact has to be something deterministic, 
>an action only a user would do by himself, like sending an INVITE, 
>MESSAGE or subscribing to a new buddy for example.

I think it would be quite impossible to determine the correct user behavior on 
the server side. Example:
- My SIP IM client (softphone) should have a high priority. I will send 
messages from it, it is active in that way. I won't call from it.
- My SIP hardphone has a lower priority. It supports PUBLISH and is active 
sending INVITE regularly.
- Only in case of calls, the BUSY triggered by my SIP hard phone's PUBLISH 
should set me to busy. This should override the softphone's presence state.
- Now, I am in a long conference. How to override it back (I set the service 
not to send messages when busy), i.e., set the IM client as available and 
override the phone state that is higher.
This is a problem.

Maybe this scenario is too particular, but I personally see the efforts of 
touching SIMPLE presence and implementing some complex state model too high for 
only some "work-around".

Either a customer does not care, or is an "early adopter" who wants to use it 
as he likes (comparing to XMPP what we all know about).
A server side implementation, if OpenSIPS logic will be changed, should then 
allow sth. per customer with different options (not per module parameter what 
should be used to determine). This will be quite complex... Some web front-end 
could help to set all this (somehow extend the presence from each client with 
rich attributes that are lacking and acc. these rules create the overall 
state.). I personally don't see another way or any hard-configured settings 
that satisfy more customers than they would disturb...

Sebastian

> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Thursday, 01. April 2010 14:13
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] SIP Presence Aggregation Issue
> 
> Hi Sebastian,
> 
> Ideally, that is correct - I agree that this is something to be done by
> the UAC.
> 
> But the problem I see is that the UACs are not able to handle multiple
> presence information (per contact) and because of this, the watcher may
> get some really confusing view over the other party.
> 
> Like if A has 2 presence-enabled devices, one publishing Busy, another
> one jsut Online, each re-publishing after 10 minutes, a watcher will
> alternatively receive NOTIFYs with BUSY and ONLINE - so you it will see
> a continue toggling between BUSY and ONLINE at each 5 minutes.
> 
> And mainly the question is - until the UAC will do a proper job, does it
> make sense to address this issue on server side, to, at least, normalize
> the viewing and get some consistent info  ?
> 
> Regards,
> Bogdan
> 
> Schumann Sebastian wrote:
> > Hi
> >
> > I think the topic touches client-side implementations more. On the 
> > server-side, it
> is the best to distribute all information and leave it up to the client to 
> interpret it.
> This is also how XMPP does it.
> >
> >
> >> -Original Message-
> >> From: users-boun...@lists.opensips.org [mailto:users-
> >> boun...@lists.opensips.org] On Behalf Of Saúl Ibarra Corretgé
> >> Sent: Thursday, 01. April 2010 13:30
> >> To: OpenSIPS users mailling list
> >> Subject: Re: [OpenSIPS-Users] SIP Presence Aggregation Issue
> >>
> >> Hi Anca,
> >>
> >> On 1/4/10 10:02 AM, Anca Vamanu wrote:
> >>
> >>> Hi all,
> >>>
> >>> This mail is a request for your comments about presence aggregation.
> >>> What is your opinion about this? Do you consider it to be a real problem
> >>> and which could be the solutions?
> >>>
> >>> Although it is possible to have more clients registering for the same
> >>> account and publishing presence information, there are not so many
> >>> clients ( none that I know of in fact ) that are able to show more th

Re: [OpenSIPS-Users] SIP Presence Aggregation Issue

[Schumann Sebastian]

Hi

I think the topic touches client-side implementations more. On the server-side, 
it is the best to distribute all information and leave it up to the client to 
interpret it. This is also how XMPP does it.

> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Saúl Ibarra Corretgé
> Sent: Thursday, 01. April 2010 13:30
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] SIP Presence Aggregation Issue
> 
> Hi Anca,
> 
> On 1/4/10 10:02 AM, Anca Vamanu wrote:
> > Hi all,
> >
> > This mail is a request for your comments about presence aggregation.
> > What is your opinion about this? Do you consider it to be a real problem
> > and which could be the solutions?
> >
> > Although it is possible to have more clients registering for the same
> > account and publishing presence information, there are not so many
> > clients ( none that I know of in fact ) that are able to show more than
> > one presence state per buddy. So, even if the presence server does
> > aggregate all the presence information for a contact and sends it to its
> > watchers, they will see only one presence state, most likely the one in
> > the top most record.
> 
> I think this happens because of how SIMPLE was initially designed. The
> presence status is undefined if different UAs provide different
> information. XMPP uses the concept of a resource, and each one has a
> different state. XMPP clients do aggregate the presence status by
> setting it to the most 'restrictive' one (if you are online + away your
> global state is away). But still, you're able to see each resources
> presence.

Talking about XMPP resources, the "overall" state shown by the program is also 
determined by it itself.

The difference to SIP is that the client has the states and priorities from all 
resources available. Usually, the highest resource is responsible for the 
overall state shown by the client (e.g., one resource priority 10 online, 
another one resource priority 20 away: User state indicated by client is away 
(orange buddy icon)).  It is however usually still possible to get the states 
from all resources that are not invisible.

The difference is that the full JID distributes the state of one presence 
source. Multiple clients, multiple full JIDs, multiple presence states with 
different priorities. The actual implementation shows the overall state of the 
bare JID usually in the buddy list.

SIP has only the URI (w/o indicating or differing between the endpoints) in the 
PIDF  element. 

However, you can put more information inside the element: s and 
priorities could be used to determine the "highest" prioritized SIP endpoint 
for communication. RPID and DM could further separate several SIP endpoints. 
Two things are required: SIP clients need to publish all this information and 
they need to interpret it and generate the overall acc. to it in the buddy list.

The problem is that for e.g., the case in RFC 3863 (4.3.1) fixed states set 
(e.g., permanent mail set with XCAP pidf-manipulation) could have a higher 
priority than automatic states. The example shows mail over IM contact (as IM 
device is busy). The correct derivation is that the user wants to be contacted 
by mail preferably and so far correct. However, I would consider it not wanted 
by the user to show him available in the buddy list (because usually a double 
click would start IM then and not mail). But this would be wrong acc. 
priorities...

Summarizing, I think if the clients provide correctly published information, it 
is up to the presence server to distribute it and the client to interpret it. 
This is my understanding in XMPP and should be similar in SIP.

Sebastian 
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Re: [OpenSIPS-Users] install siptrace

[Schumann Sebastian]

Hi

You need to use the siptrace module in OpenSIPS.
http://www.opensips.org/html/docs/modules/devel/siptrace.html

It will store the messages in a DB.

If your purpose it to display them, the control panel 
(http://opensips-cp.sourceforge.net/) should offer that.

Sebastian 

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Darshak Modi
> Sent: Thursday, 17. December 2009 13:13
> To: 'OpenSIPS users mailling list'
> Subject: [OpenSIPS-Users] install siptrace
> 
> Hi,
> Where to download this tool? 
> 
> Please provide a link .
> 
> 
> --Darshak
> 
> 
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Re: [OpenSIPS-Users] IMC Commands

[Schumann Sebastian]

Indiver,

X-Lite might sent those commands not in plain text, but HTML embedded.

The _very first_ character in the body of the message must be a sharp (i.e., 
#). As HTML starts <, it is not considered as a command.

Try using sipsak or some client where you can force plain text to create the 
channel.

Hope that helps.

Sebastian

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Indiver
> Sent: Tuesday, 10. November 2009 07:52
> To: users@lists.opensips.org
> Subject: [OpenSIPS-Users] IMC Commands
> 
> [...]
> But the problem is there is no chat room created and none of 
> the following
> imc commands are working from x-lite. But i can succesfully 
> chat between to
> two sip xlite phones.
> 
>  #create chat-000 private
> #join sip:chat-...@opensips.org  
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Re: [OpenSIPS-Users] Mediaproxy receives only

[Schumann Sebastian]

Hi Dan 

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Dan Pascu
> Sent: Thursday, 10. September 2009 01:09
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Mediaproxy receives only
> >
> > What could be a possible cause of mediaproxy (current src) to only  
> > receive media from both parties but not to relay it.
> 
> There are a number of possible causes:
> 
> 1. Check that there is no iptables rule blocking the traffic
> 2. Make sure IP forwarding is enabled:
> /proc/sys/net/ipv4/ip_forward should be 1
> 3. Make sure rp_filter is disabled:
> /proc/sys/net/ipv4/conf/all/rp_filter is 0
> /proc/sys/net/ipv4/conf/default/rp_filter is 0
> /proc/sys/net/ipv4/conf/eth0/rp_filter is 0 (replace eth0 with  
> whatever interface you use)

The rp_filter has been set to 1. I have overlooked this.

Thanks for the helpful hint!

Best regards
Sebastian
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[OpenSIPS-Users] Mediaproxy receives only

[Schumann Sebastian]

Hi

What could be a possible cause of mediaproxy (current src) to only receive 
media from both parties but not to relay it.

This is dispatcher output:
media-dispatcher[32573]: debug: Issuing "update" command to relay at 
172.16.230.2
media-dispatcher[32573]: debug: Issuing "update" command to relay at 
172.16.230.2
media-dispatcher[32573]: debug: Issuing "remove" command to relay at 
172.16.230.2
media-dispatcher[32573]: debug: Got statistics: {'from_tag': 'lfizf', 
'dialog_id': '2065:505037704', 'start_time': 1252528738.319, 'timed_out': 
False, 'call_id': 'nonhfmzknhro...@sebastian', 'to_tag': '2437949283', 
'streams': [{'status': 'closed', 'caller_codec': 'G711a', 'post_dial_delay': 
10.5756189823, 'callee_codec': 'G711a', 'start_time': 0, 'caller_bytes': 
412800, 'callee_bytes': 549600, 'caller_packets': 2064, 'end_time': 55, 
'callee_remote': 'clientip:54624', 'caller_remote': 'clientip:57592', 
'media_type': 'audio', 'callee_local': mediaproxyip:50118', 'timeout_wait': 0, 
'caller_local': 'mediaproxyip:50116', 'callee_packets': 2748}], 'duration': 55, 
'to_uri': 'al...@example.com', 'from_uri': 'b...@example.com', 'callee_ua': 
'unknown agent', 'caller_ua': 'Twinkle/1.4.2'}

This is the relay output:
media-relay[22329]: debug: Received new SDP offer
media-relay[22329]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 
50116
media-relay[22329]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 
50117
media-relay[22329]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 
50118
media-relay[22329]: mediaproxy.mediacontrol.StreamListenerProtocol starting on 
50119
media-relay[22329]: debug: Added new stream: (audio) clientip:57593 (RTP: 
Unknown, RTCP: Unknown) <-> mediaproxyip:50116 <-> mediaproxyip:50118 <-> 
Unknown (RTP: Unknown, RTCP: Unknown)
media-relay[22329]: debug: created new session nonhfmzknhro...@sebastian: 
b...@example.com (lfizf) --> al...@example.com
media-relay[22329]: debug: updating existing session nonhfmzknhro...@sebastian: 
b...@example.com (lfizf) --> al...@example.com
media-relay[22329]: debug: Received updated SDP answer
media-relay[22329]: debug: Got initial answer from callee for stream: (audio) 
clientip:57593 (RTP: Unknown, RTCP: Unknown) <->mediaproxyip:50116 <-> 
mediaproxyip:50118 <-> clientip:54624 (RTP: Unknown, RTCP: Unknown)
media-relay[22329]: debug: Got traffic information for stream: (audio) 
clientip:57593 (RTP: Unknown, RTCP: clientip:57593) <-> mediaproxyip:50116 <-> 
mediaproxyip:50118 <-> clientip:54624 (RTP: Unknown, RTCP: Unknown)
media-relay[22329]: debug: Got traffic information for stream: 
(audio)clientip:57593 (RTP: clientip:57592, RTCP: clientip:57593) 
<->mediaproxyip:50116 <-> mediaproxyip:50118 <->clientip:54624 (RTP: Unknown, 
RTCP: Unknown)
media-relay[22329]: debug: Got traffic information for stream: (audio) 
clientip:57593 (RTP: clientip:57592, RTCP:clientip:57593) <-> 
mediaproxyip:50116 <-> mediaproxyip:50118 <-> clientip:54624 (RTP: 
clientip:54624, RTCP: Unknown)
media-relay[22329]: debug: removing session nonhfmzknhro...@sebastian: 
b...@example.com (lfizf) --> al...@example.com
media-relay[22329]: (Port 50116 Closed)
media-relay[22329]: (Port 50117 Closed)
media-relay[22329]: (Port 50118 Closed)
media-relay[22329]: (Port 50119 Closed)

Traffic goes from both clients to the mediaproxy, but the mediaproxy only sends 
one packet in the beginning to both clients and then not again...

Dispatcher is colocated with OpenSIPS and relay is on a separate machine.

Any further info needed?

Thanks!

Best regards
Sebastian
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Re: [OpenSIPS-Users] mailing list reply-to : list or sender

[Schumann Sebastian]

I second that!

Sebastian


From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jeff Pyle
Sent: Wednesday, 05. August 2009 16:41
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] mailing list reply-to : list or sender

Yep.  Seems straight-forward to me.


On 8/5/09 10:03 AM, "Darren Sessions"  wrote:

I agree completely.

 - Darren


On Aug 5, 2009, at 7:58 AM, Brett Nemeroff wrote:

I agree with you here. Even if it's "different" than the way most of your other 
lists work, "Poster" simply makes more sense. And I don't think another way 
should be picked because the poster might do the wrong thing. I think it's up 
to all of us to reply to the list as appropriate.

For me, If I'm reading a message from a specific person, and I hit "reply" I'm 
replying to that person. If I hit "reply to all" I'm replying to everyone. This 
just makes sense to me.


And if you set to "The List", it's very hard to reply to the poster. Since 
neither reply to reply-all will get you the email of the poster.
-Brett


 On Tue, Aug 4, 2009 at 5:41 PM, Dan Pascu  wrote:

On Wednesday 05 August 2009, Bogdan-Andrei Iancu wrote:
 > "normal" is a fuzzy word. I can do the change, not a problem, but just
 > to past what mailman says:
 >
 > "Where are replies to list messages directed? Poster is *strongly*
 > recommended for most mailing lists.  ->  Poster This list
 > Explicit address "
 >
 > Currently is "Poster"(which is recommended) and you suggest "This List".
 >
 > To be honest I consider logical the "Poster" setting as primarily I'm
 > talking with a person on the list and not to the list.
 >
 > But any other opinions are welcome here.


"The List" is a worse choice than "Poster" because it limits my choice of whom
 I want to reply to and makes it more complicated to reply to the sender alone.
 When using "The List", if I want to reply to the sender alone, I need to
 employ manual copy/paste of his address, after I manually delete the list
 address which was automatically added when replying. This is awkward and
 annoying.

 With "Poster", I can easily use reply, to reply to the sender alone, or "reply
 all" to send to both the poster and the list and the list will make sure the
 poster will receive a single copy anyway.

 I don't think nowadays it could be an issue to anyone, as I can't imagine a
 mail user agent without a "Reply all" function.

 --
 Dan


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Re: [OpenSIPS-Users] cannot connect to xmlrpc server

[Schumann Sebastian]

Hi

In case you verify all of Bogdans proposals, check if SELINUX is enabled. 

I had once this annoying problem (under CentOS though) and it took a while till 
I figured. Even local communication with the XMLRPC interface was not possible 
in my case. 

Best regards
Sebastian 

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of 
> Bogdan-Andrei Iancu
> Sent: Friday, 29. May 2009 10:11
> To: Цэвээндорж ЖиМэйл
> Cc: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] cannot connect to xmlrpc server
> 
> Hi Tseveen,
> 
> 1) is your opensips using the mi_xmlrpc module ?
> 
> 2) check with netstat on the opensips server to see if indeed 
> mi_xmlrpc 
> is listening (netstat -tal)
> 
> 3) be sure you are trying to connect on the proper pprt 
> (mi_xmlrpc uses 
> by default 8080)
> 
> 4) firewall between ?
> 
> Regards,
> Bogdan
> 
> Цэвээндорж ЖиМэйл wrote:
> > Hello,
> >
> > I have package OpenSIPS-CP 2.0 on Ubuntu 9.04.
> >
> > I got below error when I press "Apply changes to server" 
> and Mi Command.
> >
> > sorry -- cannot connect to xmlrpc server
> > 111 - Connection refused
> >
> > What I missed configuration ?
> >
> > Sincerely,
> > Tseveen.
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
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Re: [OpenSIPS-Users] OpenSips and RLS

[Schumann Sebastian]

Check the source of the page in the browser, it might not display the tags in 
the visible output.

Sebastian

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Michael Ciupka
> Sent: Tuesday, 28. April 2009 12:56
> To: Anca Vamanu; users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] OpenSips and RLS
> 
> Hello,
> 
> oh no. thank you. i read the text now. i have another question.
> when i enter this command in the webbrowser i get always the 
> same output:
> 
> http://192.168.2.6:8000/xcap-root/pres-rules/users/alice/pres-
> rules.xml
> http://192.168.2.6:8000/xcap-root/pres-rules/users/alice/pres-
> rules.xml
> http://192.168.2.6:8000/xcap-root/pres-rules/users/alice/pres-rules
> 
> --> "allow"
> why i dont get the xml document?
> 
> Regards
> Michael
> 
>  Original-Nachricht 
> > Datum: Tue, 28 Apr 2009 13:47:31 +0300
> > Von: Anca Vamanu 
> > An: Michael Ciupka 
> > CC: users@lists.opensips.org
> > Betreff: Re: [OpenSIPS-Users] OpenSips and RLS
> 
> > Have you read the documentation :  
> http://www.opensips.org/Resources/Rls?
> > 
> > Michael Ciupka wrote:
> > > Hello,
> > >
> > > someone has a functioning main routing logic from 
> Opensips where RLS is
> > configure. opensips work well and the modparam from RLS 
> too, but i dont
> > know what code i need i the main routing logic to make RLS 
> 100% functional.
> > >
> > > it would be great if someone could help me
> > >
> > > Regards
> > > Michael
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> > >   
> 
> ___
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Re: [OpenSIPS-Users] XMLRPC

[Schumann Sebastian]

Pls. check the module docs.

The dependencies are described here:
http://www.opensips.org/html/docs/modules/1.5.x/mi_xmlrpc.html

You should remove the mi_xmlrpc from exclude_modules in the Makefile after 
satisfying the deps and compile again.

The module should be available then.

If you do not compile it yourself, check the package for the accordant distro. 
I spotted that at least for debian, it is also available as .deb.

Sebastian

PS: Please keep the topic on the list.



From: Michael Ciupka
Sent: Fri 24-Apr-09 19:46
To: Schumann Sebastian
Subject: Re: RE: [OpenSIPS-Users] XMLRPC


hello

how i compile these files?


 Original-Nachricht 
> Datum: Fri, 24 Apr 2009 19:38:42 +0200
> Von: Schumann Sebastian 
> An: Michael Ciupka , Bogdan-Andrei Iancu 
> , users@lists.opensips.org
> Betreff: RE: [OpenSIPS-Users] XMLRPC

> Did you compile the module?
> 
> Due to external dependencies it is not compiled by default.
> 
> Sebastian
> 
> 
> 
> 
> From: Michael Ciupka
> Sent: Fri 24-Apr-09 19:17
> To: Bogdan-Andrei Iancu; users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] XMLRPC
> 
> 
> Hello,
> 
> fine. I write the mi_xml in the opensips.cfg and the loadmodule
> "mi_xmlrpx.so". now i get the error in the console:
> r...@chillazz-desktop:/usr/lib/opensips/modules# /etc/init.d/opensips
> restart
> Not starting opensips: invalid configuration file!
> -e 
> Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching
> mi_xmlrpc found
> |Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config
> file, line 186, column 22-23: Can't set module parameter
> Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching
> mi_xmlrpc found
> |Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config
> file, line 187, column 20-21: Can't set module parameter
> Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching
> mi_xmlrpc found
> |Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config
> file, line 188, column 19-20: Can't set module parameter
> Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching
> mi_xmlrpc found
> |Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config
> file, line 189, column 22-23: Can't set module parameter
> Apr 24 19:15:39 [8] ERROR:core:main: bad config file (4 errors)
> 
>  Original-Nachricht 
> > Datum: Fri, 24 Apr 2009 19:54:40 +0300
> > Von: Bogdan-Andrei Iancu 
> > An: Michael Ciupka 
> > CC: users@lists.opensips.org
> > Betreff: Re: [OpenSIPS-Users] XMLRPC
> 
> > Hi Michael,
> > 
> > OpenSIPS has a built in XMLRPC server - see the mi_xmlrpc module:
> > http://www.opensips.org/html/docs/modules/devel/mi_xmlrpc.html
> > 
> > Regards,
> > Bogdan
> > 
> > Michael Ciupka wrote:
> > > Hello,
> > >
> > > i want to configure my OpenSIPS Control Panel. Now i have configure
> the
> > monit and it works well. My question is which XMLRPC Server i must
> install
> > to have the complete functionality from the OpenSIPS Control Panel.
> > > Have Opensips a XMLRPC Server implemented?
> > >
> > > Regards
> > > Michael
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >
> > >   
> 
> ___
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Re: [OpenSIPS-Users] XMLRPC

[Schumann Sebastian]

Did you compile the module?

Due to external dependencies it is not compiled by default.

Sebastian




From: Michael Ciupka
Sent: Fri 24-Apr-09 19:17
To: Bogdan-Andrei Iancu; users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] XMLRPC


Hello,

fine. I write the mi_xml in the opensips.cfg and the loadmodule "mi_xmlrpx.so". 
now i get the error in the console:
r...@chillazz-desktop:/usr/lib/opensips/modules# /etc/init.d/opensips restart
Not starting opensips: invalid configuration file!
-e 
Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching 
mi_xmlrpc found
|Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config file, 
line 186, column 22-23: Can't set module parameter
Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching 
mi_xmlrpc found
|Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config file, 
line 187, column 20-21: Can't set module parameter
Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching 
mi_xmlrpc found
|Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config file, 
line 188, column 19-20: Can't set module parameter
Apr 24 19:15:39 [8] ERROR:core:set_mod_param_regex: no module matching 
mi_xmlrpc found
|Apr 24 19:15:39 [8] CRITICAL:core:yyerror: parse error in config file, 
line 189, column 22-23: Can't set module parameter
Apr 24 19:15:39 [8] ERROR:core:main: bad config file (4 errors)

 Original-Nachricht 
> Datum: Fri, 24 Apr 2009 19:54:40 +0300
> Von: Bogdan-Andrei Iancu 
> An: Michael Ciupka 
> CC: users@lists.opensips.org
> Betreff: Re: [OpenSIPS-Users] XMLRPC

> Hi Michael,
> 
> OpenSIPS has a built in XMLRPC server - see the mi_xmlrpc module:
> http://www.opensips.org/html/docs/modules/devel/mi_xmlrpc.html
> 
> Regards,
> Bogdan
> 
> Michael Ciupka wrote:
> > Hello,
> >
> > i want to configure my OpenSIPS Control Panel. Now i have configure the
> monit and it works well. My question is which XMLRPC Server i must install
> to have the complete functionality from the OpenSIPS Control Panel.
> > Have Opensips a XMLRPC Server implemented?
> >
> > Regards
> > Michael
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   

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Re: [OpenSIPS-Users] About new new project of a middle applicationenabling opensips modules

[Schumann Sebastian]

Hi Matteo

Like this: http://www.sipwise.at/index.php/products?start=3 ?

Sebastian 

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of 
> mmarzu...@interfree.it
> Sent: Tuesday, 10. February 2009 18:47
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] About new new project of a 
> middle applicationenabling opensips modules
> 
> 
> Hi.
> Yes I'm looking for something to generate Opensips config, 
> but first I would understand if it is feasible.
> Any suggestion will be appreciated.
> 
> Thanks.
> 
> Matteo Marzuola
> 
> 
> >Hi Alex,
> 
> >I think Matteo is looking for something to generate OpenSIPS 
> config file 
> >and not a simple web interface to add users..
> 
> >Regards,
> >Bogdan
> 
> >Alex Balashov wrote:
> > http://siremis.asipto.com/
> >
> > mmarzu...@interfree.it wrote:
> >
> >   
> >> Hi all.
> >> I'm considering the possibility of achieving a middle 
> application between a client who needs to configure a certain 
> scenario in the opensips.cfg and OpenSIPS.
> >> The idea is to use a web application that allows you to 
> choose which modules to load, enter the appropriate 
> parameters and enable the appropriate routes in the script. 
> Someone has a suggestion or can suggest a project with similar goals?
> >>
> >> Thanks a lot for your support.
> >>
> >> Marzuola matteo
> >>
> >>
> >> 
> --
> --
> >> Vuoi essere presente online? 
> >> Vuoi dare voce alla tua attivita`? 
> >> Acquista un dominio su domini.interfree.it.
> >> A partire da 18,59 euro
> >> 
> --
> --
> >>
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >> 
> >
> >
> >   
> 
> 
> 
> 
> --
> --
> Vuoi essere presente online? 
> Vuoi dare voce alla tua attivita`? 
> Acquista un dominio su domini.interfree.it.
> A partire da 18,59 euro
> --
> --
> 
> 
> ___
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Re: [OpenSIPS-Users] On the edge of a new major release

[Schumann Sebastian]

Hi Alex, Bogdan

I think it might also be out of context - only use this parameter to make AOR 
not mandatory in ONREPLY_ROUTE.

As example, I wanna explain the case I use the function: I check in the request 
of the caller is registered (could do that like before AOR already) and in the 
reply the same check for the callee. I think the context in checking the caller 
(that the $T_ruri contains) by default in the reply might not be justified and 
throwing an error for that case might be the better choice. Just thought about 
it on my particular case, maybe there are other opinions out there...

Best regards
Sebastian 

> -Original Message-
> From: users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] On Behalf Of 
> Bogdan-Andrei Iancu
> Sent: Wednesday, 04. February 2009 10:31
> To: Alex Hermann
> Cc: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] On the edge of a new major release
> 
> Hi Alex,
> 
> You mean, if no AOR pushed from script, to use the RURI of 
> the branch? 
> it will be interesting, but it will create unwanted 
> dependences between 
> the REGISTER and TM module. More or less, the REGISTER module will 
> depend on TM and we shouldn't do that. But you can simply do:
> if ( registered("location","$T_ruri") )
> 
> Regards,
> Bogdan
> 
> Alex Hermann wrote:
> > On Tuesday 03 February 2009 10:58:04 Bogdan-Andrei Iancu wrote:
> >   
> >> Now you can find the answers to all this question is this 
> comprehensive
> >> compilation of OpenSIPS evolution from 1.4.0 to 1.5.0:
> >> http://www.opensips.org/index.php?n=Main.Ver15x
> >> 
> >
> > 1.16  TM modulew PV 
> > $T_ruri - the RURI of the current branch; this information 
> is taken from the 
> > transaction structure, so you can access this information 
> for any sip message 
> > (request/reply) that has a transaction.
> >
> > 1.20  REGISTRAR module
> > registered() function can be used from onreply route, 
> branch route and local 
> > route. NOTE: if used from onreply route it is mandatory to 
> pass as parameter 
> > an AOR (as replies do not have RURI).
> >
> >
> > Can't the above features be combined to lift the mentioned 
> limitation for the 
> > AOR in onreply_route.
> >   
> 
> 
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Re: [OpenSIPS-Users] On the edge of a new major release

[Schumann Sebastian]

Hi Alex

AOR is used to pass *any* AOR as parameter to the registered function for a 
check. It does not necessarily have something to do with the transaction.

Sebastian


-Original Message-
From: users-boun...@lists.opensips.org on behalf of Alex Hermann
Sent: Tue 2/3/2009 18:44
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] On the edge of a new major release
 
On Tuesday 03 February 2009 10:58:04 Bogdan-Andrei Iancu wrote:
> Now you can find the answers to all this question is this comprehensive
> compilation of OpenSIPS evolution from 1.4.0 to 1.5.0:
> http://www.opensips.org/index.php?n=Main.Ver15x

1.16  TM modulew PV 
$T_ruri - the RURI of the current branch; this information is taken from the 
transaction structure, so you can access this information for any sip message 
(request/reply) that has a transaction.

1.20  REGISTRAR module
registered() function can be used from onreply route, branch route and local 
route. NOTE: if used from onreply route it is mandatory to pass as parameter 
an AOR (as replies do not have RURI).


Can't the above features be combined to lift the mentioned limitation for the 
AOR in onreply_route.
-- 
Alex Hermann

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[OpenSIPS-Users] IMC error

[Schumann Sebastian]

Hi all
 
I get an error when I include the IMC module into my OpenSIPS configuration:
 
ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 
'name' in 'field list' 
ERROR:core:db_do_query: error while submitting query 
ERROR:imc:add_from_db: failed to querry table 
ERROR:imc:mod_init: failed to get information from db 
ERROR:core:init_mod: failed to initialize module imc 
ERROR:core:main: error while initializing modules 
 
I created both IMC tables using the script and set all parameters acc. the doc 
page.
 
Any hints what I may have forgotten or is this a bug?
 
Best regards
Sebastian
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Re: [OpenSIPS-Users] Process List Question

[Schumann Sebastian]

Hi

You can check the processes in detail using the FIFO command
$ opensipsctl fifo ps
(FIFO must be enabled).

The processes are almost as Inaki described. 

In detail, it should be like this:
1. UDP WORKER (SIP) x children x interfaces
children=10
listen=10.0.0.1:5060
listen=10.0.0.2:5060
-> 20 processes finally, so children for each interface

2. TCP WORKER (SIP) x children
tcp_children=5
listen... does not matter, fixed number of processes handles it. If no 
tcp_children is set and TCP enabled, the children number is used.
--> 5 processes

3. TCP MANAGER x1

4. TIMER PROCESSES (multiple possible)

5. ATTENDANT x1

Sometimes, also modules can require additional processes.

Sebastian

-Original Message-
From: [EMAIL PROTECTED] on behalf of Iñaki Baz Castillo
Sent: Sun 11/9/2008 21:13
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Process List Question
 
El Domingo, 9 de Noviembre de 2008, Geoffrey Mina escribió:
> I just installed OpenSIPS and got everything up and running.  Can somebody
> please explain the output of "ps -ax" to me?  I don't understand why there
> are so many processes running.
>
>
> [EMAIL PROTECTED] log]# ps ax | grep opensips
> 29371 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29373 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29374 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29375 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29376 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29377 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29378 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29379 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29380 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29381 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29382 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29383 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29384 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29385 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29386 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29387 ?S  0:00 /usr/local/sbin/opensips -m 2048
> 29470 pts/0S+ 0:00 grep opensips

It depends on the number of interfaces and transport protocol (TCP/UDP) in 
which opensips is listening. If you set "children=4" and listen in eth0 for 
TCP and UDP then you will have 4*2 + main process = 9 (well, it's something 
like this but not exactly).


-- 
Iñaki Baz Castillo

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