Re: [sr-dev] [Kamailio-Users] kamailio crashes

10 Mar 2010

Dear Marius

The scenario is as follows:

1.  A Call is placed by a sip subscriber "A"
2.  kamailio forwards the call to the asterisk server
3.  Asterisk plays an IVR message on the subscriber "A", creates a new 
call to a "virtual" number which is forwarded to the kamailio server, 
and plays an ivr to this leg as well when the call is answered, then it 
connects the two calls.
4.  Kamailio translates the "virtual" number to the pstn number of 
subscriber B

I have attached a picture of the above scenario.

The modules that are loaded are:

loadmodule "db_mysql.so"
loadmodule "mi_fifo.so"
loadmodule "mi_datagram.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "uri_db.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "acc.so"
loadmodule "dispatcher.so"
loadmodule "pdt.so"
loadmodule "dialplan.so"
loadmodule "siptrace.so"
loadmodule "dialog.so"
loadmodule "sqlops.so"
loadmodule "userblacklist.so"
loadmodule "htable.so"
loadmodule "uac.so"

The config that does all the routing is :

route[10] {

        xlog("alx ------- This is Route 10 -------");

        if($rU =~ "^.*%+")
        {
                xlog("alx ------- The number contains %23 ");
                $rU = $(rU{re.subst,/^(.*)%23(.*)/\1\2/});
                #$rU = $(rU{s.unescape.user}); #It changes the %23 to # !!
                xlog("alx ------- The perl  $rU ------- ");
        }

        if($rU =~ "^.*#+")
        {
                xlog("alx ------- The number contains #");
                $rU = $(rU{re.subst,/^(.*)#(.*)/\1\2/});
                #$rU = $(rU{s.unescape.user}); #It changes the %23 to # !!
                xlog("alx ------- The perl  $rU ------- ");
        }

        if(prefix2domain("2", "0")) {

                $var(dial_grp) = $(rd{s.select,0,.}{s.int}); # Dialplan 
group prefix for routing
                $var(num_pr) = $(rd{s.select,1,.}{s.int});  # The number 
of digits that prefix has
                $var(num_translation) = $(rd{s.select,2,.}{s.int});   # 
Called number translation
                $avp(s:port_translation) = $(rd{s.select,3,.}{s.int});   
# Port number translation
                #$var(test_var) = $(rd{s.select,4,.}{s.int});   # Future 
property

                $avp(s:cust_prefix) = $(rU{s.substr,0,$var(num_pr)});
                $rU = $(rU{s.substr,$var(num_pr),0});

        xlog("alx ------- The new rU is $rU and properties $rd -------");

        if($var(num_translation) == 1)
        {
                if($sht(a=>$rU)!=null){

                      $rU = $sht(a=>$rU);
                      xlog("alx ------- Translation Done. DST num=$rU  
----------");

                } else {
                        xlog("alx ------- Translation NOT Done  
----------");
                }

                        #xlog("alx ------- We have DST number 
translation for user fU $avp(s:frm_user_name)  ----------");
                        #if(dp_translate("31", "$rU/$rU"))
                        #{
                        #        xlog("alx ------- Translation Done. DST 
num=$rU ----------");
                        #} else {
                        #        xlog("alx ------- Translation NOT Done 
----------");
                        #}
        }

        if(dp_translate("$var(dial_grp)", "$rU/$rU"))
                {
                        xlog("alx ------- The $rU and with attributes 
:$avp(s:dest) -------\n");

                        $var(i) = 0;
                        while($(avp(s:dest){s.select,$var(i),.})!="#")
                        {
                                $avp(s:dstgrp) = 
$(avp(s:dest){s.select,$var(i),.}{s.int});
                                $var(i) = $var(i) + 1;
                                xlog("alx ------- The 
avp(s:dstgrp)=$avp(s:dstgrp) var(i)=$var(i) -------");
                        }

                        # backup the username so we can use different 
prefixes
                        $avp(s:user) = $rU;

                        # select destination from first group
                                if(ds_select_domain("$avp(s:dstgrp)",
"4"))
                                {
                                        if($(ru{uri.param,prefix})!=null)
                                                {
                                                                $ru = 
"sip:" + $(ru{uri.param,prefix})  +  $avp(s:user) + "@" + $rd;

                                                } else {
                                                                $ru = 
"sip:" + $avp(s:user) + "@" + $rd;
                                                }
                                }

                                $avp(s:dstgrp) = null;
                                        xlog("alx ------- The final RURI 
is $ru ------- ");
                                        if($avp(s:port_translation) == 1)
                                        {
                                                rewriteport("5061");
                                        }
                                        t_on_failure("3");
                                        t_relay();
                                        exit;

                }

        }

}

Attached is the trace

Regards.

P.

marius zbihlei wrote:
...
  Panagiotis Skoulikaritis wrote:
  Hello Daniel

 the kamailio version is 1.5.3

 Regards

 P.  Hello,

 Can you give us more details like the sip message that generates the 
 coredump (or if every sip message received generates the core), if 
 your config does something more out of the ordinary(let's say exotic). 
 Can we reproduce it ?

 It would also be helpful if you specify the list of modules you have 
 loaded.

 Cheers,
 Marius

 Daniel-Constantin Mierla wrote:
  Hello, 
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users 

No.     Time        Source                Destination           Protocol Info
     11 3.576857    sip subscriber A            kamailio          SIP/SDP  Request: INVITE
sip:virtual number@kamailio;transport=UDP, with session description

Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: INVITE sip:virtual number@kamailio;transport=UDP SIP/2.0
        Method: INVITE
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP sip subscriber
A:5060;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: rport
        Max-Forwards: 70
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: <sip:virtual number@kamailio>;transport=UDP
            SIP to address: sip:virtual number@kamailio
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
        Content-Type: application/sdp
        User-Agent: Zoiper rev.448
        Content-Length: 245
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): Zoiper_user 0 0 IN IP4 10.0.2.46
                Owner Username: Zoiper_user
                Session ID: 0
                Session Version: 0
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: 10.0.2.46
            Session Name (s): Zoiper_user
            Connection Information (c): IN IP4 sip subscriber A
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: sip subscriber A
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 8000 RTP/AVP 8 0 3 101
                Media Type: audio
                Media Port: 8000
                Media Proto: RTP/AVP
                Media Format: ITU-T G.711 PCMA
                Media Format: ITU-T G.711 PCMU
                Media Format: GSM 06.10
                Media Format: 101
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-15
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-15
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     12 3.577798    kamailio          sip subscriber A            SIP      Status: 100
Giving a try

Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 100 Giving a try
        Status-Code: 100
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP sip subscriber
A:5060;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        To: <sip:virtual number@kamailio>;transport=UDP
            SIP to address: sip:virtual number@kamailio
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        Server: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     13 3.577893    kamailio          asterisk          SIP/SDP  Request: INVITE
sip:virtual number@asterisk, with session description

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: INVITE sip:virtual number@asterisk SIP/2.0
        Method: INVITE
        [Resent Packet: False]
    Message Header
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bKce3c.92b7a847.0
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Max-Forwards: 69
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: <sip:virtual number@kamailio>;transport=UDP
            SIP to address: sip:virtual number@kamailio
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
        Content-Type: application/sdp
        User-Agent: Zoiper rev.448
        Content-Length: 245
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): Zoiper_user 0 0 IN IP4 10.0.2.46
                Owner Username: Zoiper_user
                Session ID: 0
                Session Version: 0
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: 10.0.2.46
            Session Name (s): Zoiper_user
            Connection Information (c): IN IP4 sip subscriber A
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: sip subscriber A
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 8000 RTP/AVP 8 0 3 101
                Media Type: audio
                Media Port: 8000
                Media Proto: RTP/AVP
                Media Format: ITU-T G.711 PCMA
                Media Format: ITU-T G.711 PCMU
                Media Format: GSM 06.10
                Media Format: 101
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-15
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-15
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     14 3.578452    asterisk          kamailio          SIP      Status: 100 Trying

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 100 Trying
        Status-Code: 100
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bKce3c.92b7a847.0
            Received: kamailio
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        To: <sip:virtual number@kamailio>;transport=UDP
            SIP to address: sip:virtual number@kamailio
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Contact: <sip:virtual number@asterisk>
            Contact Binding: <sip:virtual number@asterisk>
                URI: <sip:virtual number@asterisk>
                    SIP contact address: sip:virtual number@asterisk
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     15 3.578736    asterisk          kamailio          SIP/SDP  Status: 183 Session
Progress, with session description

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 183 Session Progress
        Status-Code: 183
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bKce3c.92b7a847.0
            Received: kamailio
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Contact: <sip:virtual number@asterisk>
            Contact Binding: <sip:virtual number@asterisk>
                URI: <sip:virtual number@asterisk>
                    SIP contact address: sip:virtual number@asterisk
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4184
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 16620
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     16 3.578818    asterisk          kamailio          SIP/SDP  Status: 200 OK, with
session description

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.0;received=kamailio
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bKce3c.92b7a847.0
            Received: kamailio
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Contact: <sip:virtual number@asterisk>
            Contact Binding: <sip:virtual number@asterisk>
                URI: <sip:virtual number@asterisk>
                    SIP contact address: sip:virtual number@asterisk
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4185 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4185
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 16620
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     17 3.579216    kamailio          sip subscriber A            SIP/SDP  Status: 183
Session Progress, with session description

Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 183 Session Progress
        Status-Code: 183
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Contact: <sip:virtual number@asterisk>
            Contact Binding: <sip:virtual number@asterisk>
                URI: <sip:virtual number@asterisk>
                    SIP contact address: sip:virtual number@asterisk
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4184
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 16620
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     18 3.580413    kamailio          sip subscriber A            SIP/SDP  Status: 200 OK,
with session description

Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5897fbd5ac888c82-1---d8754z-
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 INVITE
            Sequence Number: 1
            Method: INVITE
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Contact: <sip:virtual number@asterisk>
            Contact Binding: <sip:virtual number@asterisk>
                URI: <sip:virtual number@asterisk>
                    SIP contact address: sip:virtual number@asterisk
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4185 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4185
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 16620 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 16620
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     19 3.811307    sip subscriber A            kamailio          SIP      Request: ACK
sip:virtual number@asterisk

Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: ACK sip:virtual number@asterisk SIP/2.0
        Method: ACK
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP sip subscriber
A:5060;branch=z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-;rport
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Branch: z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-
            RPort: rport
        Max-Forwards: 70
        Route: <sip:kamailio;lr;ftag=fe71df57;did=9f7.5f04d887>
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 ACK
            Sequence Number: 1
            Method: ACK
        User-Agent: Zoiper rev.448
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     20 3.812537    kamailio          asterisk          SIP      Request: ACK sip:virtual
number@asterisk

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: ACK sip:virtual number@asterisk SIP/2.0
        Method: ACK
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bKce3c.92b7a847.2
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bKce3c.92b7a847.2
        Via: SIP/2.0/UDP sip subscriber A:5060;received=sip subscriber
A;branch=z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-;rport=5060
            Transport: UDP
            Sent-by Address: sip subscriber A
            Sent-by port: 5060
            Received: sip subscriber A
            Branch: z9hG4bK-d8754z-5463cc5518cdcc67-1---d8754z-
            RPort: 5060
        Max-Forwards: 69
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP to address: sip:virtual number@kamailio
            SIP tag: as416d507c
        From: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP from address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 1 ACK
            Sequence Number: 1
            Method: ACK
        User-Agent: Zoiper rev.448
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     42 11.624423   asterisk          kamailio          SIP/SDP  Request: INVITE
sip:virtual number@kamailio, with session description

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: INVITE sip:virtual number@kamailio SIP/2.0
        Method: INVITE
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: rport
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>
            SIP to address: sip:virtual number@kamailio
        Contact: <sip:sip subscriber A@asterisk>
            Contact Binding: <sip:sip subscriber A@asterisk>
                URI: <sip:sip subscriber A@asterisk>
                    SIP contact address: sip:sip subscriber A@asterisk
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        User-Agent: Asterisk PBX
        Max-Forwards: 70
        Date: Tue, 09 Mar 2010 17:30:11 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4184
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 13698 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 13698
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     43 11.625328   kamailio          asterisk          SIP      Status: 100 Giving a try

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 100 Giving a try
        Status-Code: 100
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>
            SIP to address: sip:virtual number@kamailio
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Server: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     44 11.625386   kamailio          PSTN        SIP/SDP  Request: INVITE
sip:6937630910@PSTN, with session description

Internet Protocol, Src: kamailio (kamailio), Dst: PSTN (PSTN)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: INVITE sip:6937630910@PSTN SIP/2.0
        Method: INVITE
        [Resent Packet: False]
    Message Header
        Record-Route: <sip:kamailio;lr=on;ftag=as59e5678f;did=631.33197276>
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK2b67.85f739f3.0
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>
            SIP to address: sip:virtual number@kamailio
        Contact: <sip:sip subscriber A@asterisk>
            Contact Binding: <sip:sip subscriber A@asterisk>
                URI: <sip:sip subscriber A@asterisk>
                    SIP contact address: sip:sip subscriber A@asterisk
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        User-Agent: Asterisk PBX
        Max-Forwards: 69
        Date: Tue, 09 Mar 2010 17:30:11 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
        Supported: replaces
        Content-Type: application/sdp
        Content-Length: 285
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 4184 4184 IN IP4 asterisk
                Owner Username: root
                Session ID: 4184
                Session Version: 4184
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: asterisk
            Session Name (s): session
            Connection Information (c): IN IP4 asterisk
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: asterisk
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 13698 RTP/AVP 3 0 8 101
                Media Type: audio
                Media Port: 13698
                Media Proto: RTP/AVP
                Media Format: GSM 06.10
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: 101
            Media Attribute (a): rtpmap:3 GSM/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 3
                MIME Type: GSM
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): silenceSupp:off - - - -
                Media Attribute Fieldname: silenceSupp
                Media Attribute Value: off - - - -
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): sendrecv

No.     Time        Source                Destination           Protocol Info
     45 11.626223   PSTN        kamailio          SIP      Status: 100 Trying

Internet Protocol, Src: PSTN (PSTN), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 100 Trying
        Status-Code: 100
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK2b67.85f739f3.0
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>
            SIP to address: sip:virtual number@kamailio
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     46 11.626916   PSTN        kamailio          SIP      Status: 503 Service
Unavailable

Internet Protocol, Src: PSTN (PSTN), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 503 Service Unavailable
        Status-Code: 503
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK2b67.85f739f3.0
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK384c143a
            RPort: 5060
        Record-Route: <sip:kamailio;lr=on;ftag=as59e5678f;did=631.33197276>
        To: <sip:virtual number@kamailio>;tag=3477144611-878554
            SIP to address: sip:virtual number@kamailio
            SIP tag: 3477144611-878554
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
SUBSCRIBE, PRACK
        Contact: <sip:6937630910@PSTN:5060>
            Contact Binding: <sip:6937630910@PSTN:5060>
                URI: <sip:6937630910@PSTN:5060>
                    SIP contact address: sip:6937630910@PSTN:5060
        Call-Info:
<sip:PSTN>;method="NOTIFY;Event=telephone-event;Duration=1000"
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     47 11.627296   kamailio          PSTN        SIP      Request: ACK
sip:6937630910@PSTN

Internet Protocol, Src: kamailio (kamailio), Dst: PSTN (PSTN)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: ACK sip:6937630910@PSTN SIP/2.0
        Method: ACK
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK2b67.85f739f3.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK2b67.85f739f3.0
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        To: <sip:virtual number@kamailio>;tag=3477144611-878554
            SIP to address: sip:virtual number@kamailio
            SIP tag: 3477144611-878554
        CSeq: 102 ACK
            Sequence Number: 102
            Method: ACK
        Max-Forwards: 70
        User-Agent: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     48 11.627838   kamailio          asterisk          SIP      Status: 444 No more tries
for you!

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 444 No more tries for you!
        Status-Code: 444
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63
            SIP to address: sip:virtual number@kamailio
            SIP tag: 0076135696eb5d2fff122699f03f5620-de63
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Server: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     49 11.628045   asterisk          kamailio          SIP      Request: ACK sip:virtual
number@kamailio

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: ACK sip:virtual number@kamailio SIP/2.0
        Method: ACK
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: rport
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63
            SIP to address: sip:virtual number@kamailio
            SIP tag: 0076135696eb5d2fff122699f03f5620-de63
        Contact: <sip:sip subscriber A@asterisk>
            Contact Binding: <sip:sip subscriber A@asterisk>
                URI: <sip:sip subscriber A@asterisk>
                    SIP contact address: sip:sip subscriber A@asterisk
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 ACK
            Sequence Number: 102
            Method: ACK
        User-Agent: Asterisk PBX
        Max-Forwards: 70
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     50 11.629887   asterisk          kamailio          SIP      Request: BYE sip:sip
subscriber A@sip subscriber A:5060

Internet Protocol, Src: asterisk (asterisk), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: BYE sip:sip subscriber A@sip subscriber A:5060 SIP/2.0
        Method: BYE
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK514b2ee7;rport
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK514b2ee7
            RPort: rport
        Route: <sip:kamailio;lr=on;ftag=fe71df57;did=9f7.5f04d887>
        From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP from address: sip:virtual number@kamailio
            SIP tag: as416d507c
        To: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP to address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 102 BYE
            Sequence Number: 102
            Method: BYE
        User-Agent: Asterisk PBX
        Max-Forwards: 70
        X-Asterisk-HangupCause: Normal Clearing
        X-Asterisk-HangupCauseCode: 16
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     51 11.630426   kamailio          sip subscriber A            SIP      Request: BYE
sip:sip subscriber A@sip subscriber A:5060

Internet Protocol, Src: kamailio (kamailio), Dst: sip subscriber A (sip subscriber A)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: BYE sip:sip subscriber A@sip subscriber A:5060 SIP/2.0
        Method: BYE
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK482e.f4934753.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK482e.f4934753.0
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK514b2ee7
            RPort: 5060
        From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP from address: sip:virtual number@kamailio
            SIP tag: as416d507c
        To: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP to address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 102 BYE
            Sequence Number: 102
            Method: BYE
        User-Agent: Asterisk PBX
        Max-Forwards: 69
        X-Asterisk-HangupCause: Normal Clearing
        X-Asterisk-HangupCauseCode: 16
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     52 11.712546   sip subscriber A            kamailio          SIP      Status: 200 OK

Internet Protocol, Src: sip subscriber A (sip subscriber A), Dst: kamailio (kamailio)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP kamailio;branch=z9hG4bK482e.f4934753.0
            Transport: UDP
            Sent-by Address: kamailio
            Branch: z9hG4bK482e.f4934753.0
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK514b2ee7
            RPort: 5060
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP to address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP from address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 102 BYE
            Sequence Number: 102
            Method: BYE
        User-Agent: Zoiper rev.448
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     53 11.712663   kamailio          asterisk          SIP      Status: 200 OK

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 200 OK
        Status-Code: 200
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP
asterisk:5060;received=asterisk;branch=z9hG4bK514b2ee7;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Received: asterisk
            Branch: z9hG4bK514b2ee7
            RPort: 5060
        Contact: <sip:sip subscriber A@sip subscriber A:5060>
            Contact Binding: <sip:sip subscriber A@sip subscriber A:5060>
                URI: <sip:sip subscriber A@sip subscriber A:5060>
                    SIP contact address: sip:sip subscriber A@sip subscriber A:5060
        To: "sip subscriber A"<sip:sip subscriber
A@kamailio>;transport=UDP;tag=fe71df57
            SIP Display info: "sip subscriber A"
            SIP to address: sip:sip subscriber A@kamailio
            SIP tag: fe71df57
        From: <sip:virtual number@kamailio>;transport=UDP;tag=as416d507c
            SIP from address: sip:virtual number@kamailio
            SIP tag: as416d507c
        Call-ID: ZTZlZWJkYWU2Y2Q1NTQ4NWZlZWE1ZDgwZTA3N2Q3ZjA.
        CSeq: 102 BYE
            Sequence Number: 102
            Method: BYE
        User-Agent: Zoiper rev.448
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     62 12.102418   kamailio          asterisk          SIP      Status: 444 No more tries
for you!

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 444 No more tries for you!
        Status-Code: 444
        [Resent Packet: True]
        [Suspected resend of frame: 48]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63
            SIP to address: sip:virtual number@kamailio
            SIP tag: 0076135696eb5d2fff122699f03f5620-de63
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Server: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

No.     Time        Source                Destination           Protocol Info
     73 13.102386   kamailio          asterisk          SIP      Status: 444 No more tries
for you!

Internet Protocol, Src: kamailio (kamailio), Dst: asterisk (asterisk)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Status-Line: SIP/2.0 444 No more tries for you!
        Status-Code: 444
        [Resent Packet: True]
        [Suspected resend of frame: 48]
    Message Header
        Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK384c143a;rport=5060
            Transport: UDP
            Sent-by Address: asterisk
            Sent-by port: 5060
            Branch: z9hG4bK384c143a
            RPort: 5060
        From: "sip subscriber A" <sip:sip subscriber
A@asterisk>;tag=as59e5678f
            SIP Display info: "sip subscriber A" 
            SIP from address: sip:sip subscriber A@asterisk
            SIP tag: as59e5678f
        To: <sip:virtual number@kamailio>;tag=0076135696eb5d2fff122699f03f5620-de63
            SIP to address: sip:virtual number@kamailio
            SIP tag: 0076135696eb5d2fff122699f03f5620-de63
        Call-ID: 1a0843ae04d1b2627d42fc602c7b043a@asterisk
        CSeq: 102 INVITE
            Sequence Number: 102
            Method: INVITE
        Server: Kamailio (1.5.3-notls (x86_64/linux))
        Content-Length: 0

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Re: [sr-dev] [Kamailio-Users] kamailio crashes