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SIP and proprietary epid= parameter & Asterisk as Media Gateway RRS feed

  • Question

  • There was a proprietary  “epid” parameter added to the From field in LCS 2005 SIP implementation.

    Result => interop problems with IP phones and other IP PBX systems.

    Please see,

    http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration

     

    Has it been removed in OCS 2007 ?

     

    Any experience with Asterisk as Media Gateway ?

    (OCS client --> OCS Server --> tls --> Mediation Server --> tcp --> openSER --> udp--> Asterisk --> analog/digital and VOIP SIP trunks to my TSP)

     

    Or is it possible to create VoIP SIP trunk  to my TSP directly from Mediation Server (udp!, SIP proxy registration is needed !) ?

     

    Thx

     

    Thursday, March 29, 2007 9:17 AM

Answers

  • Hi Daniel,

     

    The epid-parameter is not needed anymore for inbound calling via the Mediation Server.

     

    I haven't yet tested OCS with Asterisk.

     

    SIP proxy registration is not possible with the Mediation Server.

     

    Regards,

    Dieter

     

    Friday, April 6, 2007 10:12 AM

All replies

  • Hi Daniel,

     

    The epid-parameter is not needed anymore for inbound calling via the Mediation Server.

     

    I haven't yet tested OCS with Asterisk.

     

    SIP proxy registration is not possible with the Mediation Server.

     

    Regards,

    Dieter

     

    Friday, April 6, 2007 10:12 AM
  •  

     Could you post a manual how-to configure openSER as converter from UDP to TCP?

    Wednesday, July 4, 2007 9:02 AM
  •  

    Leave your mailaddress here, and i'll send you my ser config file

     

    Regards,

    Dieter

    Wednesday, July 4, 2007 10:22 AM
  •  

     vd@htservice.ru

    Thank you.

     

    Wednesday, July 4, 2007 1:49 PM
  • Could you send another to marcus_hooper@hotmail.com

     

    Thanks,

        -marcus

    Thursday, July 5, 2007 4:20 PM
  •  

    could you send one to rneubauer_at_itiliti.com

     

    Friday, October 19, 2007 6:25 AM
  • could you please do one last more to rsailer@viengineering.com ? thanks!
    Tuesday, October 23, 2007 9:15 PM
  •  

    Could you please send one copy to me onzi[at]ustrem.org?

     

    Thanks!

    Thursday, October 25, 2007 2:27 PM
  •  

    Hi all,

     

    I'll post here the routing logic of my openser config file (paste this into a standard config file)

     

    Code Block

    # -------------------------  request routing logic -------------------

    # main routing logic

    route{
            # initial sanity checks -- messages with
            # max_forwards==0, or excessively long requests
            if (!mf_process_maxfwd_header("10")) {
                    sl_send_reply("483","Too Many Hops");
                    exit;
            };

            if (method=="INVITE")
                   record_route();
            if (method=="BYE")
                   loose_route();


            if (src_ip==192.168.2.10) { #ip address of mediationserver
                    t_relay("udp:192.168.2.11:5060"); #ip address + port of gateway
            } else {
                    t_relay("tcp:192.168.2.10:5060"); #ip address + port of mediationserver
            };

    }

     

     

    Regards,

    Dieter

    Thursday, October 25, 2007 2:34 PM
  • Dieter, thanks for the config!

    If I am connecting OCS to SIP UDP trunks over the internet, WITHOUT a media gateway device (since we're not interfacing with a phone system), what do I set for the gateway IP? I assume that it should be the IP of the SIP trunk provider. Is that correct?
    Friday, October 26, 2007 3:11 PM
  • Hi Eric,

     

    Yes the GW ip can you replace with the ip of the SIP trunk provider.

    But maybe, you to www-authenticate against the SIP trunk provider, and that's not in my config file.

    Regs,

    Dieter 

    Monday, October 29, 2007 10:48 AM
  •  

    Hi all.

     

    Dieter, can you provide a sample config file with sip trunk provider??

     

    for example, asumming my voip provider is voip.myprovider.com and my username and ppasword are "usr" and "pass", how can i configure openser and ocs to use these account and provide my ocs users calls throught my sip provider???

     

     

    Thanks in advance.

     

    Monday, October 29, 2007 5:40 PM
  • Actually, the new asterisk Version can make Direct sip trunk over TCP with OCS mediation Server.
    Friday, January 30, 2009 1:16 AM
  •  
    While Asterisk 1.6 can talk directly to an OCS Mediation server - with OCS R1 - the on-hold function does not work correctly within the OC client - something that manipulation in the OpenSER script allows to happen.


    [Update Feb 6th 09] - I have tested OCS R2 with the OC R2 desk client, using Asterisk 1.6.0.1 & everything works fine - including on-hold.


    Paul
    Tuesday, February 3, 2009 7:56 PM
  • Hi Paul 
    I'm a little confuse, the conexion is  *->OCS R2->clients OCS  ??
    nothing to do??
    just intalled ?

    Regards
    Tuesday, September 22, 2009 9:51 PM

  • Hi,

    The connection is...

    * (separate server) -> OCS R2 Mediation Server (separate server) -> OCS R2 Server (separate server) -> desk OC Client

    It's important to note that it has to be Asterisk 1.6 for this layout because it can use sip over tcp/ip.


    You need to specify several things in the Asterisk server's SIP.CONF file for the connection to the OCS server to really make this work smoothly - with the main items being...

    transport=tcp
    deny=all
    allow=ulaw
    nat=yes


    Hope this helps...

    Paul Adams
    Wednesday, September 23, 2009 3:27 AM