Re: Users Digest, Vol 38, Issue 11

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Re: Users Digest, Vol 38, Issue 11

vivid333
   5. cannot resolve hostname (vivid333)


	#loadmodule "load_balancer.so"
	#loadmodule "dispatcher.so"

	# -----dispatcher params
	#modparam("dispatcher", "db_url", "mysql://opensips:opensipsrw@localhost/opensips")
	#modparam("dispatcher", "force_dst", 1)
	#modparam("dispatcher", "ds_ping_interval", 30)
	#modparam("dispatcher", "ds_probing_mode", 1)
	#modparam("dispatcher", "ds_probing_threshhold", 10)


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

        when I disable dispacher && load_balancer module,  Error "cannot resolve hostname XXX" disappear. it works now.
   But I still don't know the reason.
   




On 2011年09月05日 17:50, [hidden email] wrote:
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Today's Topics:

   1. relaying simple messages to all contacts (Bret McDanel)
   2. questions about intalling opensips in CentOS 5.5 (liujian)
   3. Re: ACK mangling redux - OpenSIPS 1.7 (Vlad Paiu)
   4. Re: questions about intalling opensips in CentOS 5.5 (Vlad Paiu)
   5. cannot resolve hostname (vivid333)


----------------------------------------------------------------------

Message: 1
Date: Mon, 05 Sep 2011 00:46:30 -0700
From: Bret McDanel [hidden email]
Subject: [OpenSIPS-Users] relaying simple messages to all contacts
To: [hidden email]
Message-ID: [hidden email]
Content-Type: text/plain; charset="UTF-8"

I am wanting to have opensips relay a sent message to all subscribed
users of the sending account, except to the actual client that sent the
message.  

In this way it would operate similar to Skype in that if you have 2 or
more clients logged into the same account they see bi-directional
messages, ie what your "other client" sends as well as what is received
to that account.


For example:
Alice registers from 3 different clients using identical credentials,
Alice[1], Alice[2] and Alice[3]

Alice[1] sends a message to Bob

The message should be delivered to Bob, Alice[2] & Alice[3] (but not
back to Alice[1]).


Currently when Bob sends a message to Alice, all 3 clients will get it,
so this part is working the way that is desired.




I had thought of enumerating the registrations and filtering the actual
sender and sending it back.  I am not able to get this working though.

An example of this would be helpful.


Any other ways to accomplish this would also be helpful.

Thanks



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MediaProxy do not add Relay Candidate

vivid333
Hello:

             A->opensips (+mediaproxy)-> B

             Can any one help me why MediaProxy don't add Relay
candidate? I am sure that use_media_proxy works before  opensips send
invite to B.


////////////////////////////A(pjsip) Send Invite,  B side receive Invite
include relay candidate
*********************************************************************************************
INVITE sip:[hidden email]:5060 SIP/2.0
Via: SIP/2.0/UDP
114.249.xxx.xxx:32768;rport;branch=z9hG4bKPjRJalhnJGIbBwiSvzyqUo2rgOM0ld3Wqo
Max-Forwards: 70
From: sip:+[hidden email];tag=qv4WrTe2HkdTK8U2W8Y40SgAeHIehYGR
To: sip:[hidden email]
Contact: <sip:+[hidden email]:32768;ob>
Call-ID: xxAb8TPG-d-ePhIOx1Xw6jyDjywCJDeU
CSeq: 23079 INVITE
Route: <sip:225.4.xxx.xxx:5060;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.10.0 Linux-2.6.32.35/i686/glibc-2.11
Content-Type: application/sdp
Content-Length:   629

v=0
o=- 3531703180 3531703180 IN IP4 114.246.xxx.xxx
s=pjmedia
c=IN IP4 114.246.xxx.xxx
t=0 0
a=X-nat:7
m=audio 56846 RTP/AVP 98 97 99 104 3 0 8 9 96
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ice-ufrag:025fe5d4
a=ice-pwd:72314495
a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 56846 typ srflx
raddr 10.0.2.15 rport 49367
a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 49367 typ host


////////////////////////////A(linphone). change Pjsip to linphone,  B
side receive Invite have no relay candidate
*********************************************************************************************
INVITE sip:[hidden email] SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:41338;rport;branch=z9hG4bK2713821050
From: <sip:[hidden email]>;tag=2642865755
To: "TEST" <sip:[hidden email]>
Call-ID: 1915536026
CSeq: 20 INVITE
Contact: <sip:vivid333@10.0.2.15:41338>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone
Supported: replaces, 100rel, timer, norefersub
Content-Length:   413

v=0
o=- 3531651689 3531651689 IN IP4 114.246.xxx.xxx
s=linphone
c=IN IP4 114.246.xxx.xxx
t=0 0
a=ice-ufrag:18e62b54
a=ice-pwd:5cf02fda
a=candidate:Sa00020f 1 UDP 1862270975 114.246.xxx.xxx 46909 typ srflx
raddr 10.0.2.15 rport 26836
a=candidate:Ha00020f 1 UDP 1694498815 10.0.2.15 26836 typ host
m=audio 46909 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv






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