Re: [OpenSIPS-Users] Help: presence not working.

[Friend_14309]

Hi.
Yes, I was also surprised to see that. But, Still my presence problem is
there :(
But, when I connect Ekiga client from PC which is directly connected to SIP
server, I can see the presence. But my DHCP clients are not able to see the
presence of each other.

Any suggestions where I should look to solve this problem.

On Fri, Nov 9, 2012 at 7:21 PM, Bogdan-Andrei Iancu wrote:

> **
> Hi,
>
> I do not see any ERROR messages in the log you send me :(.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 11/08/2012 07:28 PM, Friend_14309 wrote:
>
> Hi
> Sorry for the late reply. Here is the attached log file with  'debug=6'.
> Please let me know, what I am doing wrong.
>
> On Tue, Nov 6, 2012 at 8:09 PM, Bogdan-Andrei Iancu 
> wrote:
>
>>  In your opensips.cfg file, set be sure that the "debug" value is 6-
>> after changing, do a restart. You will opensips is very verbose now.
>>
>> Make the scenario that generates the error and send me the entire log
>> file (starting with opensips start).
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>>   On 11/06/2012 12:47 PM, Friend_14309 wrote:
>>
>> I will be happy do it, but, I am very much new to this and linux.
>> Can you please point in the direction how to do it?
>>
>> Thank you.
>>
>> On Tue, Nov 6, 2012 at 11:40 AM, Bogdan-Andrei Iancu > > wrote:
>>
>>>  Hi,
>>>
>>> You do not need pua (Presece User Agent) to simply do presence between
>>> end-points.
>>>
>>> Could you run opensips in full debug mode (set debug=6) and send me (off
>>> list) the entire log generated by opensips ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>
>>>   On 11/06/2012 10:32 AM, Friend_14309 wrote:
>>>
>>> Hello,
>>>
>>> No, there no NAT in my test network, just simple static routing.
>>> also, No. There is no other Error message before that "udp_send" ERROR
>>> message.
>>>
>>> One thing that I would like to ask is, I did not install "pua" modules.
>>> Are they necessary if I am implementing simple presence server i.e.
>>> everyone is allowed to see the presence of each other.
>>>
>>> Thank you.
>>>
>>> On Mon, Nov 5, 2012 at 8:36 PM, Bogdan-Andrei Iancu >> > wrote:
>>>
  Hi,

 Do you have any kind of NAT in your used network (like one of the
 clients behind a NAT) ?

 Also, before the "udp_send" ERROR, do you see any other error from the
 same process ?

 Regards,

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


 On 11/05/2012 01:06 PM, Friend_14309 wrote:

  Hi. I am new to this world, so please help.
 I am trying to set up a lab. I have 2 DHCP servers and a SIP server.

 My DHCP server ans OpenSIPS are working fine. But, when I register my
 clients to SIP server, I can not see there presence, though I can make call
 between clients.

 I get these messages,


 "INFO:presence:handle_expired_subs: notify"

 "INFO:presence:send_notify_request: NOTIFY sip:mika@172.17.100.5 via
 sip:mika@172.16.102.11 on behalf of sip:mika@172.17.100.5 for event
 presence.winfo, to_tag=b11fa29be7deff2756188f5829692e66-e5da, cseq=3"

 "ERROR:tm:msg_send: udp_send failed"
 "ERROR:tm:t_forward_nonack: sending request failed"

 *Please help. Thank you.*



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>>>
>>
>
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[OpenSIPS-Users] CRITICAL:dialog:log_next_state_dlg:

[Dragomir Haralambiev]

Hello ,

I use latest Opensips 1.8.2.
In opensips.log I see:

CRITICAL:dialog:log_next_state_dlg: bogus event 7 in state 2 for dlg
0x7f82211f3008 [409:916221583] with clid
'77296ab5684dd56e0e2660fb285c3dc9@192.168.1.111:2000' and tags 'as6a7a723d'
'3561551315-165083'

 What is the reason for this?

Best regards,
PlayMen
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[OpenSIPS-Users] Load Balancer between two OpenSIPs server. Redis Cluster [HOWTO]

[Bakko]

Hello list,

I written the first draft about Load Balancer between two Opensips 
Servers in a Redis Cluster.


I'm sorry if the HowTo is in Spanish, but my English is not very good.

http://www.voztovoice.org/?q=node/585 



Thank you Bogdan for your help and sorry if my mails arrive directly to 
your private e-mail. I thought I was writing to the list.


Regards
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Re: [OpenSIPS-Users] Load Balancer Issue

[Nilanjan Banerjee]

Hi Bogdan,

  You were spot on. Fixed the ROUTE headers for ACK and BYE in the sipp
scenario and everything just worked perfectly. Thanks a lot for bearing
with me with your continued support.  Much appreciate it.

Nilanjan.

On Fri, Nov 9, 2012 at 11:07 PM, Bogdan-Andrei Iancu wrote:

> **
> Nilanjan,
>
> The caller script is broken - in ACK, is should be ROUTE hdrs where you
> have the RECORD-ROUTE ones :) .ACK should look like:
>
>
> U 2012/11/01 11:19:02.006514 X.X.X.23:5080 -> X.X.X.206:5060
> ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
> Route:  ,
> .
> Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-5.
> From: sipp ;tag=31168SIPpTag001.
> To: sut ;tag=30500SIPpTag011.
> Call-ID: 1-31168@X.X.X.23.
> CSeq: 1 ACK.
> Contact: sip:sipp@X.X.X.23:5080.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
>
> I guess it is an err in your sipp scenario.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 11/09/2012 07:24 PM, Nilanjan Banerjee wrote:
>
> Hi Bogdan,
>
>   Thanks for your reply. The answer to both your questions is yes. Sample
> of OK and ACK at the caller as follows:
>
> #
> U 2012/11/01 11:19:02.006375 X.X.X.206:5060 -> X.X.X.23:5080
> SIP/2.0 200 OK.
> Record-Route: ,
> .
> Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-0.
> From: sipp ;tag=31168SIPpTag001.
> To: sut ;tag=30500SIPpTag011.
> Call-ID: 1-31168@X.X.X.23.
> CSeq: 1 INVITE.
> Contact: .
> Content-Type: application/sdp.
> Content-Length:   137.
> .
> v=0.
> o=user1 53655765 2353687637 IN IP4 X.X.X.5.
> s=-.
> c=IN IP4 X.X.X.5.
> t=0 0.
> m=audio 6000 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
>
> #
> U 2012/11/01 11:19:02.006514 X.X.X.23:5080 -> X.X.X.206:5060
> ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
> Record-Route:  ,
> .
> Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-5.
> From: sipp ;tag=31168SIPpTag001.
> To: sut ;tag=30500SIPpTag011.
> Call-ID: 1-31168@X.X.X.23.
> CSeq: 1 ACK.
> Contact: sip:sipp@X.X.X.23:5080.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
> .
>
> Thanks,
> Nilanjan.
>
> On Fri, Nov 9, 2012 at 10:29 PM, Bogdan-Andrei Iancu 
> wrote:
>
>>  Hi Nilanjan,
>>
>> Check in the trace if :
>> 1) the 200 OK getting back to the caller has 2 RR headers (one from Proxy
>> and one from LB).
>>
>> 2) the ACK from caller (before LB) has 2 Route headers, one pointing to
>> LB, next to Proxy.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>>   On 11/09/2012 05:01 PM, Nilanjan Banerjee wrote:
>>
>> Hi Bogdan,
>>
>>   Thanks a lot for your suggestion and sorry for the delay in getting
>> back with this...I tried the following configuration as you have suggested
>> for the Load Balancer and the Proxy:
>>
>> __
>> Load Balancer:
>> __
>>
>> route{
>> if (!mf_process_maxfwd_header("3")) {
>> sl_send_reply("483","looping");
>> exit;
>> }
>>
>> if (!has_totag()) {
>> # initial request
>> record_route();
>> } else {
>> # sequential request -> obey Route indication
>> loose_route();
>> t_relay();
>> exit;
>> }
>>
>> # detect resources and do load balancing
>>
>>  load_balance("1","sc");
>>
>> # LB function returns negative if no suitable destination (for
>> requested resources) is found,
>> # or if all destinations are full
>> if ($retcode<0) {
>>  sl_send_reply("500","Service full");
>>  exit;
>> }
>>
>> xlog("Selected destination is: $du\n");
>>
>> # send it out
>> if (!t_relay()) {
>> sl_reply_error();
>> }
>> }
>>
>> __
>> Proxy
>> __
>>
>> route{
>>
>> if (!has_totag()) {
>> # initial request
>> record_route();
>> } else {
>> # sequential request -> obey Route indication
>> loose_route();
>> }
>>
>> if (!t_relay()) {
>>  #   xlog("L_ERR","sl_reply_error\n");
>> sl_reply_error();
>> }
>>
>> }
>>
>> However, I am still getting the same error - basically the ACK and the
>> BYE messages are skipping the Proxy and the response to the BYE is sent to
>> the Proxy. Here are the sample ACK and BYE for the following setup I am
>> using:
>>
>> X.X.X.23:5080 --> X.X.X.206:5060 --> X.X.X.8:5060 --> X.X.X.5:5070
>> (sipp UAC)   --> (Load Balancer) -->  (Proxy) --> (sipp UAS)
>>
>> #
>> U 2012/11/01 11:19:22.901990 X.X.X.206:5060 -> X.X.X.5:5070
>> ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
>> Record-Route:  ,
>> .
>> Via: SIP/2.0/UDP X.X.X.206;branch=z9hG4bK0112.20fe162.2.
>> Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-5-5.
>> From: sipp ;tag=31168SIPpTag005.
>> To: sut ;tag=30500SIPpTag015.
>> Call

[OpenSIPS-Users] does opensips support rtmp

[zhi sun]

hi guys,

does opensips support rtmp? i want to use flash rtmp phone (on web page) to
call opensips (and the FS behind).

if not, what is the correct point as solution?

thanks in advance,
-iamsyt
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