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/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>
   To: "30003016094191500"<sip:30003016094191500@10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403@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>;tag=1
   To: "30003016094191500"<sip:30003016094191500@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>;tag=1
   To: "30003016094191500" <sip:30003016094191500@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>;tag=ygQBtp6QpKtcD
   From: "22222238515000403"<sip:22222238515000403@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>;tag=1
   To: "30003016094191500" <sip:30003016094191500@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>
  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





---------- 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
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