[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
Brian West
brian at freeswitch.org
Mon Aug 24 13:42:31 PDT 2009
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:
>
>
>
> 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/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>
> To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
> From: "22222238515000403"<sip:22222238515000403 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>;tag=1
> To: "30003016094191500"<sip:30003016094191500 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>;tag=1
> To: "30003016094191500" <sip:
> 30003016094191500 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>;tag=ygQBtp6QpKtcD
> From: "22222238515000403"<sip:22222238515000403 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>;tag=1
> To: "30003016094191500" <sip:
> 30003016094191500 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>
> From: "22222238515000403"<sip:22222238515000403 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
>
>
> _______________________________________________
> 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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