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<sip%3A30003016094191500@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@10.4.4.252<sip%3A22222238515000403@10.4.4.252>
To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403@10.4.4.251>
;tag=1 Call-ID: 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<sip%3A22222238515000403@10.4.4.251>
;tag=1 To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251>
Call-ID: 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<sip%3A22222238515000403@10.4.4.251>
;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251> ;tag=ygQBtp6QpKtcD Call-ID: 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<sip%3A30003016094191500@10.4.4.251>
;tag=ygQBtp6QpKtcD From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403@10.4.4.251> ;tag=1 Call-ID: 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<sip%3A22222238515000403@10.4.4.251>
;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251> ;tag=ygQBtp6QpKtcD Call-ID: 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.
---------- Forwarded message ---------- From: Brian West <brian@freeswitch.org> To: 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:
------------------------------------------------------------------------ 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<sip%3A30003016094191500@10.4.4.251>
From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403@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
---------- Forwarded message ---------- From: "Jerry Richards" <jerry.richards@teotech.com> To: <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> To: 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:
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> To: 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:
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> To: 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:
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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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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