[Opensips - B2BUA] ERROR : No dialog found

2011-12-09
2013-05-09
  • Florent Volpo
    Florent Volpo
    2011-12-09

    Hello,

    I Have a problem on Opensips, to use B2BUA with a refer scenario.

    I have this problem problem :
    ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid= , method=REFER

    When, the B2BUA receive the REFER method, the call transfer doesn't work..

    Could someone help me please ?

    Thanks in advance,

    Florent.


    My logs with this error:

    Dec  9 11:31:12 localhost opensips: DBG:core:destroy_avp_list: destroying list (nil)
    Dec  9 11:31:12 localhost opensips: DBG:core:receive_msg: cleaning up
    Dec  9 11:31:12 localhost opensips: DBG:tm:utimer_routine: timer routine:4,tl=0xb59ab8cc next=(nil), timeout=9700000
    Dec  9 11:31:12 localhost opensips: DBG:tm:utimer_routine: timer routine:4,tl=0xb59a9964 next=(nil), timeout=9800000
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_msg: SIP Request:
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_msg:  method:  <REFER>
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_msg:  uri:     <sip:10.24.246.91:5062>
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_msg:  version: <SIP/2.0>
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_headers: flags=2
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKddf1.a7cf4052.0>; state=16
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_via: end of header reached, state=5
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_headers: via found, flags=2
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_headers: this is the first via
    Dec  9 11:31:15 localhost opensips: DBG:core:receive_msg: After parse_msg…
    Dec  9 11:31:15 localhost opensips: DBG:core:receive_msg: preparing to run routing scripts…
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_headers: flags=ffffffffffffffff
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to_param: tag=13086SIPpTag011
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to_param: tag=13086SIPpTag011
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to: end of header reached, state=29
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to: display={}, ruri={sip:111@10.24.246.91:5063}
    Dec  9 11:31:15 localhost opensips: DBG:core:get_hdr_field: <To> ; uri=
    Dec  9 11:31:15 localhost opensips: DBG:core:get_hdr_field: to body
    Dec  9 11:31:15 localhost opensips: DBG:core:get_hdr_field: cseq <CSeq>: <1> <REFER>
    Dec  9 11:31:15 localhost opensips: DBG:core:get_hdr_field: content_length=0
    Dec  9 11:31:15 localhost opensips: DBG:core:get_hdr_field: found end of header
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: start - method = REFER
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: <uri> host:port
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: <socket> address:port
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: <socket> address:port
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to_param: tag=8ce09851b296d7c859872c90a73bb9de-8f38
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to: end of header reached, state=29
    Dec  9 11:31:15 localhost opensips: DBG:core:parse_to: display={}, ruri={sip:sipp@127.0.0.1:5060}
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_parse_key: Does not have b2b_entities prefix
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_parse_key: hash_index =   - local_index=
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: received a b2b client request
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_search_htable_next_dlg: searching   totag
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_search_htable_next_dlg: searching fromtag
    Dec  9 11:31:15 localhost opensips: DBG:b2b_entities:b2b_prescript_f: No dialog found
    Dec  9 11:31:15 localhost opensips: ERROR:b2b_entities:b2b_prescript_f: No dialog found, callid= , method=REFER

    Dec  9 11:31:15 localhost opensips: REQU:<REFER> cid:B2B.368.4751306 ruri:sip:10.24.246.91:5062 furi:sip:sipp@127.0.0.1:5060 turi:sip:111@10.24.246.91:5063 cseq:1  (len:438)
    Dec  9 11:31:15 localhost opensips: REFER sip:10.24.246.91:5062 SIP/2.0#015#012Via: SIP/2.0/UDP 10.24.246.91:5062;branch=z9hG4bKddf1.a7cf4052.0#015#012From: <sip:sipp@127.0.0.1:5060>;tag=8ce09851b296d7c859872c90a73bb9de-8f38#015#012To: <sip:111@10.24.246.91:5063>;tag=13086SIPpTag011;tag=13086SIPpTag011#015#012Call-ID: B2B.368.4751306#015#012CSeq: 1 REFER#015#012Contact: <sip:127.0.0.1:5063;transport=UDP>#015#012Max-Forwards: 70#015#012Subject: Performance Test#015#012Content-Length: 0#015#012Refer-To: <sip:10.24.243.11:5060>#015#012#015#012

     
  • Hi Florent,

    Do you have a sip capture for the entire scenario (both sides) along with the corresponding logs ? you can upload them on  pastebin or so.

    Regards,
    Bogdan

     
  • Florent Volpo
    Florent Volpo
    2011-12-12

    Hi Bogdan,

    Thanks you for your answer.

    I use SIPp to test a dialog with Opensips : SIPp (UAC) <---------> Opensips (BBUA) <---------> SIPp (UAS)

    Attached my opensips.cfg, uac and uas scenarios.

    ---------------------------------- opensips.cfg ----------------------------------------

    debug=6
    log_stderror=no
    log_facility=LOG_LOCAL5

    fork=yes
    children=1

    /* uncomment the following lines to enable debugging */
    #debug=6
    #fork=no
    #log_stderror=yes

    /* uncomment the next line to disable TCP (default on) */
    disable_tcp=yes

    /* uncomment the next line to enable the auto temporary blacklisting of
       not available destinations (default disabled) */
    #disable_dns_blacklist=no

    /* uncomment the next line to enable IPv6 lookup after IPv4 dns
       lookup failures (default disabled) */
    #dns_try_ipv6=yes

    /* uncomment the next line to disable the auto discovery of local aliases
       based on revers DNS on IPs (default on) */
    #auto_aliases=no

    /* uncomment the following lines to enable TLS support  (default off) */
    #disable_tls = no
    #listen = tls:your_IP:5061
    #tls_verify_server = 1
    #tls_verify_client = 1
    #tls_require_client_certificate = 0
    #tls_method = TLSv1
    #tls_certificate = "/usr/local/etc/opensips/tls/user/user-cert.pem"
    #tls_private_key = "/usr/local/etc/opensips/tls/user/user-privkey.pem"
    #tls_ca_list = "/usr/local/etc/opensips/tls/user/user-calist.pem"

    # default db_url to be used by modules requiring DB connection
    #db_default_url="mysql://opensips:opensipsrw@localhost/opensips"
    #db_default_url="mysql://root:@localhost/test"

    port=5062
    #mhomed=0

    /* uncomment and configure the following line if you want opensips to
       bind on a specific interface/port/proto (default bind on all available) */
    #listen=udp:10.24.246.228:5062

    ####### Modules Section ########

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

    /* uncomment next line for MySQL DB support */
    loadmodule "db_mysql.so"
    loadmodule "tm.so"
    loadmodule "signaling.so"
    loadmodule "sl.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 "uac_auth.so"
    loadmodule "b2b_entities.so"
    loadmodule "b2b_logic.so"
    loadmodule "dispatcher.so"
    loadmodule "dialog.so"

    /* uncomment next lines for MySQL based authentication support
       NOTE: a DB (like db_mysql) module must be also loaded */
    #loadmodule "auth.so"
    #loadmodule "auth_db.so"
    /* uncomment next line for aliases support
       NOTE: a DB (like db_mysql) module must be also loaded */
    #loadmodule "alias_db.so"
    /* uncomment next line for multi-domain support
       NOTE: a DB (like db_mysql) module must be also loaded
       NOTE: be sure and enable multi-domain support in all used modules
             (see "multi-module params" section ) */
    #loadmodule "domain.so"
    /* uncomment the next two lines for presence server support
       NOTE: a DB (like db_mysql) module must be also loaded */
    #loadmodule "presence.so"
    #loadmodule "presence_xml.so"

    # ----------- setting module-specific parameters ----------

    # --- B2BUA ---

    modparam("b2b_logic", "script_scenario", "/usr/local/etc/opensips/b2b_refer.xml")

    modparam("b2b_entities", "script_req_route", "b2b_request")
    modparam("b2b_entities", "script_reply_route", "b2b_reply")

    modparam("b2b_entities", "db_mode", 0)
    modparam("b2b_logic", "db_mode", 0)

    #modparam("b2b_logic", "db_url", "mysql://root:florent@localhost/opensips")

    #Not compulsory for script scenarios
    #modparam("b2b_logic", "server_address", "sip:10.24.246.91:5062")

    # --- tm_module ---
    modparam("tm", "pass_provisional_replies", 1)

    # --- dispatcher ---
    modparam("dispatcher", "list_file", "/usr/local/etc/opensips/dispatcher.list")

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

    # --- rr params ---
    # do not append from tag to the RR (no need for this script)
    modparam("rr", "append_fromtag", 0)

    # --- registrar params ---
    /* uncomment the next line not to allow more than 10 contacts per AOR */
    #modparam("registrar", "max_contacts", 10)

    # --- usrloc params ---
    modparam("usrloc", "db_mode",   0)
    /* uncomment the following lines if you want to enable DB persistency
       for location entries */
    #modparam("usrloc", "db_mode",   2)
    #modparam("usrloc", "db_url",
    # "mysql://opensips:opensipsrw@localhost/opensips")

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

    # --- acc params ---
    /* what sepcial events should be accounted ? */
    modparam("acc", "early_media", 1)
    modparam("acc", "report_cancels", 1)
    /* by default ww do not adjust the direct of the sequential requests.
       if you enable this parameter, be sure the enable "append_fromtag"
       in "rr" module */
    modparam("acc", "detect_direction", 0)
    /* account triggers (flags) */
    modparam("acc", "failed_transaction_flag", 3)
    modparam("acc", "log_flag", 1)
    modparam("acc", "log_missed_flag", 2)
    /* uncomment the following lines to enable DB accounting also */
    modparam("acc", "db_flag", 1)
    modparam("acc", "db_missed_flag", 2)

    # --- auth_db params ---
    /* uncomment the following lines if you want to enable the DB based
       authentication */
    #modparam("auth_db", "calculate_ha1", yes)
    #modparam("auth_db", "password_column", "password")
    #modparam("auth_db", "db_url",
    # "mysql://opensips:opensipsrw@localhost/opensips")
    #modparam("auth_db", "load_credentials", "")

    # --- alias_db params ---
    /* uncomment the following lines if you want to enable the DB based
       aliases */
    #modparam("alias_db", "db_url",
    # "mysql://opensips:opensipsrw@localhost/opensips")

    # --- domain params ---
    /* uncomment the following lines to enable multi-domain detection
       support */
    #modparam("domain", "db_url",
    # "mysql://opensips:opensipsrw@localhost/opensips")
    #modparam("domain", "db_mode", 1)   # Use caching

    # --- multi-module params ---
    /* uncomment the following line if you want to enable multi-domain support
       in the modules (dafault off) */
    #modparam("auth_db|usrloc|uri", "use_domain", 1)

    # --- presence params ---
    /* uncomment the following lines if you want to enable presence */
    #modparam("presence|presence_xml", "db_url",
    # "mysql://opensips:opensipsrw@localhost/opensips")
    #modparam("presence_xml", "force_active", 1)
    #modparam("presence", "server_address", "sip:192.168.1.2:5060")

    ####### Routing Logic ########

    # main request routing logic

    route{

            xlog("L_NOTICE","REQU:<$rm> cid:$ci ruri:$ru furi:$fu turi:$tu cseq:$cs  (len:$ml)\n");
            xlog("L_DBG", "$mb\n");
            xlog("L_DBG", "$rb\n");

    if (!mf_process_maxfwd_header("10")) {
    sl_send_reply("483","Too Many Hops");
    exit;
    }

    if (has_totag()) {
    # sequential request withing a dialog should
    # take the path determined by record-routing

    if (loose_route()) {
    if (is_method("BYE")) {
    #setflag
    }
    else if (is_method("INVITE")) {
    # even if in most of the cases is useless, do RR for
    # re-INVITEs alos, as some buggy clients do change route set
    # during the dialog.
    record_route();
    }

    xlog("loose_route");
    route(1);

    } else {
    if ( is_method("ACK") ) {
                                  if ( t_check_trans() ) {
    xlog("non loose-route, but stateful ACK; must be an ACK after");
    # non loose-route, but stateful ACK; must be an ACK after
    # a 487 or e.g. 404 from upstream server
    t_relay();
    exit;
    } else {
    xlog("ACK without matching transaction -> ignore and discard");
    # ACK without matching transaction ->
    # ignore and discard
    exit;
    }
    }
    sl_send_reply("404","Not here");
    }
    exit;
    }

    #initial requests

    # CANCEL processing
    if (is_method("CANCEL"))
    {
    if (t_check_trans())
    t_relay();
    exit;
    }

    t_check_trans();

    # preloaded route checking
    if (loose_route()) {
    xlog("L_ERR",
    "Attempt to route with preloaded Route's ");
    if (!is_method("ACK"))
    sl_send_reply("403","Preload Route denied");
    exit;
    }

    if (is_method("INVITE") && !(src_ip == "10.24.246.91" && src_port == 5062) ) {

    xlog("avant b2b_init_request");
    b2b_init_request("top hiding");
    xlog("apres b2b_init_request");
    exit;
                  
    #record_route();
    #t_relay();
    #exit;

    #if ($rU =="111"){
    # ds_select_dst("1","4");
                    #        xlog("Calling 111 !");
                    #}
    }

    if ($rU==NULL) {
    # request with no Username in RURI
    sl_send_reply("484","Address Incomplete");
    exit;
    }

    route(1);
    }

    route {
    xlog("route");
    if (!t_relay()){
    sl_reply_error();
    };
    exit;
    }

    onreply_route {

        xlog("L_NOTICE","RESP:<$rs> cid:$ci furi$fu turi:$tu cseq:$cs  (len:$ml)\n");
        xlog("L_DBG", "$mb\n");
        xlog("L_DBG", "$rb\n");

    }

    route {
      xlog("b2b_request ($ci)\n");
        xlog("L_NOTICE","b2b_request REQU:<$rm> cid:$ci furi$fu turi:$tu cseq:$cs  (len:$ml)\n");
    }

    route {
      xlog("b2b_reply ($ci)\n");
        xlog("L_NOTICE","b2b_reply RESP:<$rs> cid:$ci furi$fu turi:$tu cseq:$cs  (len:$ml)\n");
    }

    local_route {
      xlog("local_route ($ci)\n");
        xlog("L_NOTICE","local_route SENT:<$rm> cid:$ci furi$fu turi:$tu cseq:$cs  (len:$ml)\n");
    }

    ------------------------------- SIPp uac.xml ------------------------------------

    <?xml version="1.0" encoding="ISO-8859-1" ?>

    <!DOCTYPE scenario SYSTEM "sipp.dtd">

    <!- This program is free software; you can redistribute it and/or      ->

    <!- modify it under the terms of the GNU General Public License as     ->

    <!- published by the Free Software Foundation; either version 2 of the ->

    <!- License, or (at your option) any later version.                    ->

    <!-                                                                    ->

    <!- This program is distributed in the hope that it will be useful,    ->

    <!- but WITHOUT ANY WARRANTY; without even the implied warranty of     ->

    <!- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      ->

    <!- GNU General Public License for more details.                       ->

    <!-                                                                    ->

    <!- You should have received a copy of the GNU General Public License  ->

    <!- along with this program; if not, write to the                      ->

    <!- Free Software Foundation, Inc.,                                    ->

    <!- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             ->

    <!-                                                                    ->

    <!-                 Sipp default 'uac' scenario.                       ->

    <!-                                                                    ->

    <scenario name="Basic Sipstone UAC">

      <!- In client mode (sipp placing calls), the Call-ID MUST be         ->

      <!- generated by sipp. To do so, use  keyword.                ->

      <send retrans="500" start_rtd="1">

        <![CDATA[

          INVITE sip:@10.24.246.91:5063 SIP/2.0

          Via: SIP/2.0/ :;branch=

          From: sipp <sip:sipp@:>;tag=SIPpTag00

          To: sut <sip:@:>

          Call-ID:

          CSeq: 1 INVITE

          Contact: sip:sipp@:

          Max-Forwards: 70

          Subject: Performance Test

          Content-Type: application/sdp

          Content-Length:

          v=0

          o=- 3413194329 3413194329 IN IP

          s=SJphone

          c=IN IP

          t=0 0

          m=audio  RTP/AVP 8

          c=IN IP

          a=rtpmap:8 PCMA/8000

        ]]>

      </send>

      <recv response="100" optional="true" rtd="1">

      </recv>

      <recv response="180" optional="true" rtd="1">

      </recv>

      <recv response="183" optional="true" rtd="1">

      </recv>

      <!- By adding rrs="true" (Record Route Sets), the route sets         ->

      <!- are saved and used for following messages sent. Useful to test   ->

      <!- against stateful SIP proxies/B2BUAs.                             ->

      <recv response="200" start_rtd="2" rrs="true">

      </recv>

      <!- Packet lost can be simulated in any send/recv message by         ->

      <!- by adding the 'lost = "10"'. Value can be  percent.       ->

      <send rtd="2">

        <![CDATA[

          ACK  SIP/2.0

          Via: SIP/2.0/ :;branch=

          From: sipp <sip:sipp@:>;tag=SIPpTag00

          To: sut <sip:@:>

          Call-ID:

          CSeq: 1 ACK

          Contact: sip:sipp@:

          Max-Forwards: 70

          Subject: Performance Test

          Content-Length: 0

        ]]>

      </send>

      <!- This delay can be customized by the -d command-line option       ->

      <!- or by adding a 'milliseconds = "value"' option here.             ->

      <pause milliseconds="4000"/>

      <recv request="BYE" crlf="true" rtd="3">

      </recv>

      <!- definition of the response time repartition table (unit is ms)   ->

      <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

      <!- definition of the call length repartition table (unit is ms)     ->

      <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

    </scenario>

    ------------------------------- SIPp uas.xml ------------------------------------

    <?xml version="1.0" encoding="ISO-8859-1" ?>
    <!DOCTYPE scenario SYSTEM "sipp.dtd">

    <!- This program is free software; you can redistribute it and/or      ->
    <!- modify it under the terms of the GNU General Public License as     ->
    <!- published by the Free Software Foundation; either version 2 of the ->
    <!- License, or (at your option) any later version.                    ->
    <!-                                                                    ->
    <!- This program is distributed in the hope that it will be useful,    ->
    <!- but WITHOUT ANY WARRANTY; without even the implied warranty of     ->
    <!- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      ->
    <!- GNU General Public License for more details.                       ->
    <!-                                                                    ->
    <!- You should have received a copy of the GNU General Public License  ->
    <!- along with this program; if not, write to the                      ->
    <!- Free Software Foundation, Inc.,                                    ->
    <!- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             ->
    <!-                                                                    ->
    <!-                 Sipp default 'uas' scenario.                       ->
    <!-                                                                    ->

    <scenario name="Basic UAS responder">
      <!- By adding rrs="true" (Record Route Sets), the route sets         ->
      <!- are saved and used for following messages sent. Useful to test   ->
      <!- against stateful SIP proxies/B2BUAs.                             ->
      <recv request="INVITE" crlf="true" rrs="true" start_rtd="1">
      </recv>

      <!- The '' keyword is replaced automatically by the          ->
      <!- specified header if it was present in the last message received  ->
      <!- (except if it was a retransmission). If the header was not       ->
      <!- present or if no message has been received, the ''       ->
      <!- keyword is discarded, and all bytes until the end of the line    ->
      <!- are also discarded.                                              ->
      <!-                                                                  ->
      <!- If the specified header was present several times in the         ->
      <!- message, all occurences are concatenated (CRLF seperated)        ->
      <!- to be used in place of the '' keyword.                   ->

      <send rtd="1">
        <![CDATA[

          SIP/2.0 100 Trying
         
         
          ;tag=SIPpTag01
         
         
          Contact: <sip::;transport=>
          Content-Length: 0

        ]]>
      </send>

      <send rtd="1">
        <![CDATA[

          SIP/2.0 180 Ringing
         
         
          ;tag=SIPpTag01
         
         
          Contact: <sip::;transport=>
          Content-Length: 0

        ]]>
      </send>

      <send retrans="500" start_rtd="2">
        <![CDATA[

          SIP/2.0 200 OK
         
         
          ;tag=SIPpTag01
         
         
          Contact: <sip::;transport=>
          Content-Type: application/sdp
          Content-Length:
         

          v=0
          o=- 1463032673 1463032673 IN IP4 193.251.214.154
          s=sviwinf09.35
          c=IN IP4 193.251.214.154
          t=0 0
          m=audio 20070 RTP/AVP 8 101
          a=rtpmap:8 PCMA/8000
          a=rtpmap:101 telephone-event/8000
          a=fmtp:101 0-15
          a=ptime:20
          a=sendrecv

        ]]>
      </send>

      <recv request="ACK" rtd="2" crlf="true">
      </recv>

      <pause milliseconds="2000"/>

      <send rtd="2">
        <![CDATA[

          REFER  SIP/2.0
         
         
         
         
         
          Contact: <sip::;transport=>
          Refer-To: <sip:10.24.243.11:5060>
          Content-Length: 0

        ]]>
      </send>

      <recv response="202" crlf="true" rtd="2">

      </recv>

      <recv request="BYE" start_rtd="3">
      </recv>

      <send rtd="3">
        <![CDATA[

          SIP/2.0 200 OK
         
         
         
         
         
          Contact: <sip::;transport=>
          Content-Length: 0

        ]]>
      </send>

      <pause milliseconds="4000"/>

      <!- definition of the response time repartition table (unit is ms)   ->
      <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

      <!- definition of the call length repartition table (unit is ms)     ->
      <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

    </scenario>

     
  • Florent Volpo
    Florent Volpo
    2011-12-12

    You can find the tcpdump of my dialog and my corresponding log here : http://pastebin.com/kaiBvpGt

    Thanks you in advance.

    Regards,
    Florent.

     
  • Florent Volpo
    Florent Volpo
    2011-12-12

    Sorry but I paste the wrong opensips.cfg,

    In the initial INVITE, it's  : b2b_init_request("refer") and not b2b_init_request("top hiding")

     
  • Florent Volpo
    Florent Volpo
    2011-12-13

    I have resolved my error. :-)

    The problem was about the "totag" of my REFER method (send by SIPp UAS), i have changed it and now the call transfer work.

     
  • Hi Florent,

    I was about to tell you that the REFER is bogusly generated with the wrong tag, but you were faster than me :)

    Regards,
    Bogdan

     
  • Florent Volpo
    Florent Volpo
    2011-12-13

    Thanks you anyway Bogdan ;-)

    Regards,

    Florent

     
  • Florent Volpo
    Florent Volpo
    2012-01-03

    Bogdan,

    In the REFER method, the totag must be the same that the fromtag generated by the B2BUA please ?

    Regards,

    Florent.