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[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

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anthony.minessale at g...
Guest

Posted: Mon Aug 24, 2009 12:38 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Your ACK message must not be valid (dialog matching or something else)
so every 1 call will generate 30 retries that are queued up in the sip stack.

at 100cps you will be generating this problem 100 times per second and queue up countless unfinished dialogs thus
eating up the cpu.

On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

Quote:

Hello,

I've been with freeswittch for a while now.. and i can say it is worth developing it.

anyhow i got into a strange issue... I'm tryng to see what load FS on my server can take. The Call flow is like this:

SIPp                   FS

INVITE -------->
           <------- 100 Trying
           <------- 302 Moved Temporary
ACK    --------->

I use a dummy dialplan for that. All custom functions i've build are disabled and i'm not using it here. Also custom modules are not loaded as well.

   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         
         <action application="log" data="INFO ######################## ServiceLookup ########################\n"/>
         <action application="log" data="INFO ######################## contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ######################## CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ######################## RADIUS auth = '${authResult}' ##########\n"/>
         <action application="execute_extension" data="doRedirect XML public"/>
        </condition>
   </extension>

   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ######################## RADIUS auth OK!!!' ##########\n"/>
         
         
         <action application="redirect" data="sip:12345616094191500@pgw01.ot.hr:5060"/>
         
         
         <action application="hangup" data="USER_BUSY"/>
         <anti-action application="redirect" data="sip:12345616094191500@pgw01.ot.hr:5060"/>
         <anti-action application="log" data="INFO ######################## RADIUS auth NOK!! ##########\n"/>
         
         <anti-action application="hangup" data="USER_BUSY"/>
      </condition>
   </extension>

When i place a call from x-lite everything works fine ... x-lite sends an invite, gets SIP 302 and ACKs it correctly... FS is happy.

When i place a call from SIPp i have the same scenario except FS seems not understand ACK message from SIPp and re-sends SIP 302 multiple times untill it gives up.

I beleive this is due to 302 resend issue but; when i load FS with 100 CPS, i can see high CPU usage (just one thread taking most load... the rest does almost nothing) on FS. Also, starting from 40 CPS there is a big delay in receiving SIP 302 messages meaning i've sent 6000 calls and so far only for half of them got 302 response.

Does anybody have a clue ?

Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 1 -l 4000):

freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])>
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 ACK
   Contact: sip:sipp@10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

Tihomir.

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tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 2:11 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Quote:

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

---------- Forwarded message ----------
From: "Raffaele P. Guidi" <raffaele.p.guidi@gmail.com (raffaele.p.guidi@gmail.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 20:24:28 +0200
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem
Actually I did that and it worked fine. I had the problem the SECOND time I run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were not

On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot@gmail.com (carlos.talbot@gmail.com)> wrote:

Quote:

When you configure FreePBX for the first time it wipes out the sip_profiles directory. If you follow the FreePBX shortcut on your desktop it'll mention this on the last screen of the configuration. You might see something such as the following below. If you plan to use FreePBX you'll need to define trunk groups, trunks, etc in order to have the sip_profiles directory populated.

regards,

Carlos

Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!

D:/FreeSWITCH/conf/sip_profiles/external.xml
D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
D:/FreeSWITCH/conf/sip_profiles/internal.xml

On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <raffaele.p.guidi@gmail.com (raffaele.p.guidi@gmail.com)> wrote:

Quote:

I had the sweet surprise to find the installer packaged with FreePBX... really great! Why it has not been advertised as it deserves? It worked like a breeze once launched, with the automatic configuration and all of that., Only thing: once stopped I cannot get it to load sofia profiles anymore - issueing sofia status doesn't show anything. I had to copy internal.xml and default.xml from a previous installation and everything got to work again - but no FreePBX anymore Sad

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Mon Aug 24, 2009 2:22 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

In your scenario you need to add [peer_tag_param] at the end of the to
on the Ack.

/b

On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:

Quote:

------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:

------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To: "30003016094191500"<sip:30003016094191500@10.4.4.251>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
Call-ID: 1-6962@10.4.4.252
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 3:42 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Hello Brian,

it doesn't work .. tried this today as well:

freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 20:28:09.367300:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])>
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
   Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
   CSeq: 1 ACK
   Contact: sip:sipp@10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
   To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

This thing is driving me crazy, pls help.

T.

Quote:

---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org (brian@freeswitch.org)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 14:15:40 -0500
Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
In your scenario you need to add [peer_tag_param] at the end of the to on the Ack.

/b

On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:

Quote:

------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

---------- Forwarded message ----------
From: "Jerry Richards" <jerry.richards@teotech.com (jerry.richards@teotech.com)>
To: <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Date: Mon, 24 Aug 2009 12:24:42 -0700
Subject: [Freeswitch-users] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]
Hello All,

I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide. I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:

"The person at extension 1000 is not available..."

Also, the Freeswitch log shows:

Cannot create outgoing channel type [error] cause:
[FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
[FACILITY_NOT_SUBSCRIBED]

Does anyone know why I get this error?

Best Regards,
Jerry

---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org (brian@freeswitch.org)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 14:33:22 -0500
Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]
Are you trying to test everything on the same machine?

/b

On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:

Quote:

Hello All,

I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide. I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:

"The person at extension 1000 is not available..."

Also, the Freeswitch log shows:

Cannot create outgoing channel type [error] cause:
[FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
[FACILITY_NOT_SUBSCRIBED]

Does anyone know why I get this error?

Best Regards,
Jerry

---------- Forwarded message ----------
From: Michael Jerris <mike@jerris.com (mike@jerris.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 15:44:18 -0400
Subject: Re: [Freeswitch-users] Problem with cnam.js?
Every page on the wiki should be editable. If you don't already have an account, go to:

http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup

Mike

On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:

Quote:

I think there’s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js.

If you use it as is, it displays “Content-type: text/html” for the effective_caller_id_name. In cnam.pl, the first two output lines are generated by:

if (!$debug) {print "Content-type: text/html\n\n";}

with the actual name in the third line.

So I changed:

fd.open("read");
buff = fd.readln();

if(buff) {
   logger(buff, "info");
   session.setVariable("effective_caller_id_name", buff);
}

To:

fd.open("read");
buff = fd.readAll();

if(buff[2]) {
   logger(buff, "info");
   session.setVariable("effective_caller_id_name", buff[2]);
}

Or remove the print statement from cnam.pl.

Sorry for the code, but the page was not editable.

Lars

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---------- Forwarded message ----------
From: Michael Jerris <mike@jerris.com (mike@jerris.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 15:46:58 -0400
Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
Do you have an answer in the dialplan for that extension? Also, check out the ignore_early_media variable.

Mike

On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:

Quote:

Hi,

I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe 3.4.4 drivers. But now I have another problem...
I want to originate calls through event socket, and I only want to receive ANSWERED(+OK) reply when the user actually answers.

Now the situation is:

====================================
originate openzap/1/a/123456 023
2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT: CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci=[0000000000]
2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer OpenZAP/1:1/123456!
API CALL [originate(openzap/1/a/123456 023)] output:
+OK f8fca2be-8fa7-11de-9076-511e29dfc082

2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer OpenZAP/1:1/123456 to XML[023@default]
freeswitch@emo-voip> 2009-08-23 08:44:06.743475 [INFO] mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel [OpenZAP/1:1/123456] has been answered
2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2 (OpenZAP/1:1/123456) Ended
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close Channel OpenZAP/1:1/123456 [CS_DESTROY]
====================================

Extension 023 is an IVR. As you can see FreeSWITCH answers the call (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel [OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).

So Sangoma drivers/daemons report the events correctly.
How can I set FreeSWITCH to answer after receiving RX EVENT (N): CALL_ANSWERED from the driver?

Thank you,
V. Panayotov
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brian at freeswitch.org
Guest

Posted: Mon Aug 24, 2009 3:48 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

ACK [url=sip:[service]@[remote_ip]:[remote_port]]sip:[service]@[remote_ip]:[remote_port][/url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <[url=sip:sipp@[local_ip]:[local_port]]sip:sipp@[local_ip]:[local_port][/url]>;tag=[call_number]
To: sut <[url=sip:[service]@[remote_ip]:[remote_port]]sip:[service]@[remote_ip]:[remote_port][/url]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: [url=sip:sipp@[local_ip]:[local_port]]sip:sipp@[local_ip]:[local_port][/url]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Use that.. your scenario has some hard coded IP's in the fields that shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

Quote:

Hello Brian,

it doesn't work .. tried this today as well:

freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 20:28:09.367300:
------------------------------------------------------------------------
INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
Max-Forwards: 70
Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])>
To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
CSeq: 1 INVITE
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 131

v=0
o=user1 53655765 2353687637 IN IP4 10.4.4.252
s=-
c=IN IP4 10.4.4.252
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

------------------------------------------------------------------------
recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
To: "30003016094191500" <sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>;tag=ygQBtp6QpKtcD
Call-ID: 1-7019@10.4.4.252 (1-7019@10.4.4.252)
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

This thing is driving me crazy, pls help.

T.

Quote:

------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To: "30003016094191500"<sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email])>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251 ([email]sip%3A22222238515000403@10.4.4.251[/email])>;tag=1
Call-ID: 1-6962@10.4.4.252 (1-6962@10.4.4.252)
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Quote:

I think there’s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js.

If you use it as is, it displays “Content-type: text/html” for the effective_caller_id_name. In cnam.pl, the first two output lines are generated by:

if (!$debug) {print "Content-type: text/html\n\n";}

with the actual name in the third line.

So I changed:

fd.open("read");
buff = fd.readln();

if(buff) {
logger(buff, "info");
session.setVariable("effective_caller_id_name", buff);
}

To:

fd.open("read");
buff = fd.readAll();

if(buff[2]) {
logger(buff, "info");
session.setVariable("effective_caller_id_name", buff[2]);
}

Or remove the print statement from cnam.pl.

Sorry for the code, but the page was not editable.

Lars

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---------- Forwarded message ----------
From: Michael Jerris <mike@jerris.com (mike@jerris.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Date: Mon, 24 Aug 2009 15:46:58 -0400
Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
Do you have an answer in the dialplan for that extension? Also, check out the ignore_early_media variable.

Mike

On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:

Quote:

dave at 3c.co.uk
Guest

Posted: Mon Aug 24, 2009 3:48 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Hi Tihomir -

I'm no SIP guru, but the things which look suspicious about the ACK to
me are:
- Via header - different branch
- Contact header - differs from INVITE

--Dave

Quote:

Hi Anthony,

I'm aware it is generating 30 retries per a call and this is killing
me ...

I lost my entire working day to figure out what is missing in the damn
ACK message SIPp is sending back... ACK looks quite ok to me.

pls can you help ?

freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
16:44:26.527236:
------------------------------
------------------------------------------
INVITE sip:30003016094191500@10.4.4.251 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
Max-Forwards: 70
Contact: <sip:22222238515000403@10.4.4.252>
To: "30003016094191500"<sip:30003016094191500@10.4.4.251>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 131

v=0
o=user1 53655765 2353687637 IN IP4 10.4.4.252
s=-
c=IN IP4 10.4.4.252
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000

------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:

------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"<sip:30003016094191500@10.4.4.251>
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:

------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From: "22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"
<sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:

------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To: "30003016094191500"<sip:30003016094191500@10.4.4.251>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
Call-ID: 1-6962@10.4.4.252
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

What m'i missing ?

Your ACK message must not be valid (dialog matching or
something else)
so every 1 call will generate 30 retries that are queued up in
the sip stack.

at 100cps you will be generating this problem 100 times per
second and queue up countless unfinished dialogs thus
eating up the cpu.

On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga
<tculjaga@gmail.com> wrote:
Hello,

I've been with freeswittch for a while now.. and i can
say it is worth developing it.

anyhow i got into a strange issue... I'm tryng to see
what load FS on my server can take. The Call flow is
like this:

SIPp FS

INVITE -------->
<------- 100 Trying
<------- 302 Moved Temporary
ACK --------->

I use a dummy dialplan for that. All custom functions
i've build are disabled and i'm not using it here.
Also custom modules are not loaded as well.

<extension name="ServiceLookup">
<condition field="destination_number"
expression="(^300030)(.*)">

<action application="log" data="INFO
######################## ServiceLookup
########################\n"/>
<action application="log" data="INFO
######################## contact = '${red_contact}'
##############\n"/>
<action application="log" data="INFO
######################## CallerNum =
'${caller_id_number:6:16}' ##########\n"/>
<action application="log" data="INFO
######################## RADIUS auth = '${authResult}'
##########\n"/>
<action application="execute_extension"
data="doRedirect XML public"/>
</condition>
</extension>

<extension name="doRedirect">
<condition field="destination_number"
expression="^doRedirect$"/>
<condition field="${authResult}" expression="^0
$|^60$">
<action application="log" data="INFO
######################## RADIUS auth OK!!!' ##########
\n"/>


<action application="redirect"
data="sip:12345616094191500@pgw01.ot.hr:5060"/>


<action application="hangup"
data="USER_BUSY"/>
<anti-action application="redirect"
data="sip:12345616094191500@pgw01.ot.hr:5060"/>
<anti-action application="log" data="INFO
######################## RADIUS auth NOK!! ##########
\n"/>

<anti-action application="hangup"
data="USER_BUSY"/>
</condition>
</extension>

When i place a call from x-lite everything works
fine ... x-lite sends an invite, gets SIP 302 and ACKs
it correctly... FS is happy.

When i place a call from SIPp i have the same scenario
except FS seems not understand ACK message from SIPp
and re-sends SIP 302 multiple times untill it gives
up.

I beleive this is due to 302 resend issue but; when i
load FS with 100 CPS, i can see high CPU usage (just
one thread taking most load... the rest does almost
nothing) on FS. Also, starting from 40 CPS there is a
big delay in receiving SIP 302 messages meaning i've
sent 6000 calls and so far only for half of them got
302 response.

Does anybody have a clue ?

Here is a trace taken on FS for calls originated from
SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 100 -trace_msg
-inf test.txt -m 1 -l 4000):

freeswitch@l01sipindir1> recv 573 bytes from
udp/[10.4.4.252]:5060 at 16:44:26.527236:

------------------------------------------------------------------------
INVITE sip:30003016094191500@10.4.4.251 SIP/2.0
Via: SIP/2.0/UDP
10.4.4.252;branch=z9hG4bK-6962-1-0;rport
Max-Forwards: 70
Contact: <sip:22222238515000403@10.4.4.252>
To:
"30003016094191500"<sip:30003016094191500@10.4.4.251>
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 131

v=0
o=user1 53655765 2353687637 IN IP4 10.4.4.252
s=-
c=IN IP4 10.4.4.252
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000

------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at
16:44:26.527566:

------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To:
"30003016094191500"<sip:30003016094191500@10.4.4.251>
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at
16:44:26.535582:

------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP
10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"
<sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info,
sla, include-session-description, presence.winfo,
message-summary, refer
Content-Length: 0

------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at
16:44:26.535809:

------------------------------------------------------------------------
ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
Via: SIP/2.0/UDP
10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
To:
"30003016094191500"<sip:30003016094191500@10.4.4.251>
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
Call-ID: 1-6962@10.4.4.252
CSeq: 1 ACK
Contact: sip:sipp@10.4.4.252:5060
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at
16:44:27.037070:

------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP
10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"
<sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info,
sla, include-session-description, presence.winfo,
message-summary, refer
Content-Length: 0

------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at
16:44:28.037063:

------------------------------------------------------------------------
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP
10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
From:
"22222238515000403"<sip:22222238515000403@10.4.4.251>;tag=1
To: "30003016094191500"
<sip:30003016094191500@10.4.4.251>;tag=Hr4mHDUeBSNyH
Call-ID: 1-6962@10.4.4.252
CSeq: 1 INVITE
Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info,
sla, include-session-description, presence.winfo,
message-summary, refer
Content-Length: 0

Tihomir.

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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

---------- Forwarded message ----------
From: "Raffaele P. Guidi" <raffaele.p.guidi@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 24 Aug 2009 20:24:28 +0200
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows
installer - great but I have a little problem
Actually I did that and it worked fine. I had the problem the
SECOND time I run FS and freepbx. And (@Brian) mod_sofia was
loaded but sip_profiles were not

On Sun, Aug 16, 2009 at 16:04, Carlos Talbot
<carlos.talbot@gmail.com> wrote:
When you configure FreePBX for the first time it wipes
out the sip_profiles directory. If you follow the
FreePBX shortcut on your desktop it'll mention this on
the last screen of the configuration. You might see
something such as the following below. If you plan to
use FreePBX you'll need to define trunk groups,
trunks, etc in order to have the sip_profiles
directory populated.

regards,

Carlos

Incompatible Configuration
WARNING: THE FOLLOWING FILES WILL BE DELETED!

* D:/FreeSWITCH/conf/sip_profiles/external.xml
* D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
* D:/FreeSWITCH/conf/sip_profiles/internal.xml

On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi
<raffaele.p.guidi@gmail.com> wrote:

I had the sweet surprise to find the installer
packaged with FreePBX... really great! Why it
has not been advertised as it deserves? It
worked like a breeze once launched, with the
automatic configuration and all of that., Only
thing: once stopped I cannot get it to load
sofia profiles anymore - issueing sofia status
doesn't show anything. I had to copy
internal.xml and default.xml from a previous
installation and everything got to work again
- but no FreePBX anymore Sad

I'm sure I'm
missing something important.

Can you give me a hint? Should sofia profiles
be served by curl or something?

Thanks,
Raffaele

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--
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave@3c.co.uk
W: http://www.3c.co.uk

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tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 5:18 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Hello Brian, Dave Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed branch within ACK message to meet INVITE's one....but it is still not enough... Also i checked RFC and this is how should it be ... (ACK without contact taking care to have correct TAGs and branch)... what can it be? ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])> From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079@10.4.4.252 (1-7079@10.4.4.252) CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079@10.4.4.252 (1-7079@10.4.4.252) CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 (1-7079@10.4.4.252) CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 (1-7079@10.4.4.252) CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 (1-7079@10.4.4.252) CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 here is a scenario i use: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC"> <send retrans="500"> <![CDATA[ INVITE sip:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="302" rtd="true"> </recv> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ]]> </send> <!-- 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> ---------- Forwarded message ---------- From: Brian West <brian@freeswitch.org (brian@freeswitch.org)> To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org) Date: Mon, 24 Aug 2009 15:42:31 -0500 Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Use that.. your scenario has some hard coded IP's in the fields that shouldn't be there. /b On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote: Hello Brian, it doesn't work .. tried this today as well:

anthony.minessale at g...
Guest

Posted: Mon Aug 24, 2009 5:27 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both. If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance? -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email]) IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email]) iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email]) pstn:213-799-1400

tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 5:35 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l 4000
scenario file:      uac_redirect.xml
FS dialplan:       public.xml
SIP trace:          trace.log

Here it is... sorry for not including at first...

T.

On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

Quote:

What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use.
please send both.

If we have to stop what we are doing to prove this works are you prepared to offer your soul to help
document and other project maintenance?

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
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tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 5:54 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

... documentation hate that Smile

) ... but thats my life actually... thats what i do for living Smile

of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS.

well ... my soul? .. is it really necessary?

T.

On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

Quote:

anthony.minessale at g...
Guest

Posted: Mon Aug 24, 2009 5:59 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]

shouldn't that be

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]

On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

Quote:

... documentation hate that Smile

) ... but thats my life actually... thats what i do for living Smile

Quote:

tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 6:21 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0

http://sipp.sourceforge.net/doc3.0/reference.html

[branch] - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message.

http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se branch being same for INVITE, 302 and ACK messages...

[branch-3] makes branch of ACK message same as INVITE....

here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd message...

freeswitch@l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at 23:11:34.506681:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])>
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403@10.4.4.252:5060 [7797f5ae-9103-11de-95e3-cda626a03c4b]
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500@pgw01.ot.hr:5060)
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
   From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n)
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute hangup(USER_BUSY)
2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/22222238515000403@10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403@10.4.4.252:5060) Ended
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403@10.4.4.252:5060 [CS_DESTROY]
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

Quote:

... documentation hate that Smile

) ... but thats my life actually... thats what i do for living Smile

Quote:

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
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tculjaga at gmail.com
Guest

Posted: Mon Aug 24, 2009 6:25 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

Is there any way to make FS complain about what header is wrong ?

T.

On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

Quote:

this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0

http://sipp.sourceforge.net/doc3.0/reference.html

[branch] - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message.

http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se branch being same for INVITE, 302 and ACK messages...

[branch-3] makes branch of ACK message same as INVITE....

here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd message...

freeswitch@l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at 23:11:34.506681:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500@10.4.4.251 ([email]sip%3A30003016094191500@10.4.4.251[/email]) SIP/2.0

   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403@10.4.4.252 ([email]sip%3A22222238515000403@10.4.4.252[/email])>
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>

   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------

send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying

   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>

   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

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

2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403@10.4.4.252:5060 [7797f5ae-9103-11de-95e3-cda626a03c4b]
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500@pgw01.ot.hr:5060)
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily

   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1

   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

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

recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0

   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
   From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1

   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

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

2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n)
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute hangup(USER_BUSY)
2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/22222238515000403@10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403@10.4.4.252:5060) Ended
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403@10.4.4.252:5060 [CS_DESTROY]
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily

   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1

   To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093@10.4.4.252 (1-7093@10.4.4.252)
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500@pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

Quote:

... documentation hate that Smile

) ... but thats my life actually... thats what i do for living Smile

Quote:

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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mayamatakeshi at gmail...
Guest

Posted: Mon Aug 24, 2009 9:00 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:

Quote:

The Via definition in your SIPp scenario differs between the INVITE and the ACK:

INVITE:
Via: SIP/2.0/[transport] [local_ip];branch=[branch]

ACK:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]

In the INVITE, you are not adding the [local_port] as you do in the ACK.
Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.
So it seems FS is not checking just the ACK's branch against the
INVITE's; it seems it is checking the whole Via header.
I don't know if this is in accordance to SIP specs.

Another thing, about the way you are calling SIPp: do no use "-sn uac"
and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
means "use the internal (embedded) scenario named xxx". So this
conflicts with the other parameter "-sf" which specifies an external
profile. It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.

regards,
takeshi

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mayamatakeshi at gmail...
Guest

Posted: Mon Aug 24, 2009 9:02 pm Post subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand

On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:

Quote:

On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:

Quote:

I mean, an external scenario (file).

It seems this doesn't cause any problem (probably because in

Quote:

the sipp startup, -sf overrides -sn), but it is misleading.

regards,
takeshi

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