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

Tihomir Culjaga tculjaga at gmail.com
Mon Aug 24 13:37:24 PDT 2009


Hello Brian,

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



freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
20:28:09.367300:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 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 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-7019 at 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 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at 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 at 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 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 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 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at 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.



>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at 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:
>
>
>>   ------------------------------------------------------------------------
>> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>>   ------------------------------------------------------------------------
>>   ACK sip:30003016094191500 at 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 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>   Call-ID: 1-6962 at 10.4.4.252
>>   CSeq: 1 ACK
>>   Contact: sip:sipp at 10.4.4.252:5060
>>   Max-Forwards: 70
>>   Subject: Performance Test
>>   Content-Length: 0
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jerry Richards" <jerry.richards at teotech.com>
> To: <freeswitch-users at 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 at freeswitch.org>
> To: freeswitch-users at 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:
>
>  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 at jerris.com>
> To: freeswitch-users at 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:
>
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at 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:
>
>  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 at default]
>> freeswitch at 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
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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