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/2.0
Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
Max-Forwards: 70
Contact: <sip:[email protected]>
To: "30003016094191500"<sip:[email protected]>
From: "22222238515000403"<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]>;tag=1
To: "30003016094191500"<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]>;tag=1
To: "30003016094191500" <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]>;tag=ygQBtp6QpKtcD
From: "22222238515000403"<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]>;tag=1
To: "30003016094191500" <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]>
From: "22222238515000403"<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
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---------- 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
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