Hello Brian, it doesn't work .. tried this today as well:
freeswi...@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 20:28:09.367300: ------------------------------------------------------------------------ INVITE sip:[email protected]<sip%[email protected]>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport Max-Forwards: 70 Contact: <sip:[email protected]<sip%[email protected]> > To: "30003016094191500"<sip:[email protected]<sip%[email protected]> > From: "22222238515000403"<sip:[email protected]<sip%[email protected]> >;tag=1 Call-ID: [email protected] 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:[email protected]<sip%[email protected]> >;tag=1 To: "30003016094191500"<sip:[email protected]<sip%[email protected]> > Call-ID: [email protected] 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:[email protected]<sip%[email protected]> >;tag=1 To: "30003016094191500" <sip:[email protected]<sip%[email protected]> >;tag=ygQBtp6QpKtcD Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]: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:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport To: "30003016094191500"<sip:[email protected]<sip%[email protected]> >;tag=ygQBtp6QpKtcD From: "22222238515000403"<sip:[email protected]<sip%[email protected]> >;tag=1 Call-ID: [email protected] CSeq: 1 ACK Contact: sip:[email protected]: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:[email protected]<sip%[email protected]> >;tag=1 To: "30003016094191500" <sip:[email protected]<sip%[email protected]> >;tag=ygQBtp6QpKtcD Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]: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 <[email protected]> > To: [email protected] > 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:[email protected]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport >> To: >> "30003016094191500"<sip:[email protected]<sip%[email protected]> >> > >> From: >> "22222238515000403"<sip:[email protected]<sip%[email protected]> >> >;tag=1 >> Call-ID: [email protected] >> CSeq: 1 ACK >> Contact: sip:[email protected]:5060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> > > > > > > ---------- Forwarded message ---------- > From: "Jerry Richards" <[email protected]> > To: <[email protected]> > 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 <[email protected]> > To: [email protected] > 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 <[email protected]> > To: [email protected] > 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 > [email protected] > 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 <[email protected]> > To: [email protected] > 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[...@default] >> freeswi...@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 >> [email protected] >> 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 > [email protected] > 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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