New subject: 2 SIP Lines, Difficulty Originating Call

8 Sep 2007

      I attempted the session.streamFile(...) and was able to successfully 
hear the wav file.  I also attempted calling the session.answer() for a 
second time within the cf_bridge_call function.  Still, I am unable to 
hear the ringing sound even though the phone that I called is actually 
ringing.  When the phone is answered, the originate appears to fail due 
to a 60 second timeout, which is reported as a NO_ANSWER.  This happens 
regardless of whether or not the phone I am calling is answered or not.

Here is the current execution flow within my Javascript:

function cf_bridge_call(phone) {
    var str = "sofia/gateway/line2/"+phone;
    session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)");
    session.preAnswer(); /* attemped a session.answer() here in another 
test.  The sound file still played, not no ringing was heard.  */
    session.streamFile("/usr/local/freeswitch/sounds/you_rang.wav");
    session.execute("bridge", str);
}

session.answer();
session.flushDigits();
/**
 * Query database and dynamically load a phone number into variable
 **/
var phoneNo = db_get_phone();
cf_bridge_call(phoneNo);

The log output was:

2007-09-08 01:54:35 [DEBUG] switch_ivr_play_say.c:1007 
switch_ivr_play_file() done playing file
2007-09-08 01:54:35 [NOTICE] switch_channel.c:385 
switch_channel_set_name() New Chan sofia/sip.vonics.net/18881234567 
[1f766c60-5de9-11dc-af73-3b3c22977fdb]
2007-09-08 01:54:35 [DEBUG] mod_sofia.c:1431 sofia_outgoing_channel() 
sofia/sip.vonics.net/18881234567 State Change CS_NEW -> CS_INIT
2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 
switch_core_session_signal_state_change() Kill 
sofia/sip.vonics.net/18881234567 [BREAK]
2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:383 
switch_core_session_run() (sofia/sip.vonics.net/18881234567) State INIT
2007-09-08 01:54:35 [DEBUG] mod_sofia.c:92 sofia_on_init() SOFIA INIT
2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event 
[nua_i_state] status [0][INVITE sent] session: 
sofia/sip.vonics.net/18881234567
2007-09-08 01:54:35 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() 
Channel sofia/sip.vonics.net/18881234567 entering state [calling]
2007-09-08 01:54:35 [DEBUG] mod_sofia.c:105 sofia_on_init() 
sofia/sip.vonics.net/18881234567 State Change CS_INIT -> CS_RING
2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 
switch_core_session_signal_state_change() Kill 
sofia/sip.vonics.net/18881234567 [BREAK]
2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:418 
switch_core_session_run() (sofia/sip.vonics.net/18881234567) State RING
2007-09-08 01:54:35 [DEBUG] mod_sofia.c:120 sofia_on_ring() SOFIA RING
2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:48 
originate_on_ring() sofia/sip.vonics.net/18881234567 State Change 
CS_RING -> CS_HOLD
2007-09-08 01:54:35 [DEBUG] switch_core_session.c:647 
switch_core_session_signal_state_change() Kill 
sofia/sip.vonics.net/18881234567 [BREAK]
2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:596 
switch_core_session_run() (sofia/sip.vonics.net/18881234567) State HOLD
2007-09-08 01:54:35 [DEBUG] switch_core_state_machine.c:204 
switch_core_standard_on_hold() Standard HOLD
2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:619 
switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 
20ms
2007-09-08 01:54:35 [DEBUG] switch_ivr_originate.c:667 
switch_ivr_originate() Play Ringback Tone [%(2000, 4000, 440.0, 480.0)]
2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event 
[nua_r_invite] status [401][Unauthorized] session: 
sofia/sip.vonics.net/18881234567
2007-09-08 01:54:35 [DEBUG] sofia_reg.c:691 
sofia_reg_handle_sip_r_challenge() Authenticating 'FreeSWITCH' with 
'Digest:"sip.vonics.net":16191111111:5874'.
2007-09-08 01:54:35 [DEBUG] sofia.c:71 sofia_event_callback() event 
[nua_i_state] status [0][INVITE sent] session: 
sofia/sip.vonics.net/18881234567
2007-09-08 01:54:35 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() 
Channel sofia/sip.vonics.net/18881234567 entering state [calling]
2007-09-08 01:55:35 [NOTICE] switch_ivr_originate.c:805 
switch_ivr_originate() Hangup sofia/sip.vonics.net/18881234567 [CS_HOLD] 
[NO_ANSWER]
2007-09-08 01:55:35 [DEBUG] switch_channel.c:1076 
switch_channel_perform_hangup() Kill sofia/sip.vonics.net/18881234567 [KILL]
2007-09-08 01:55:35 [DEBUG] switch_core_session.c:647 
switch_core_session_signal_state_change() Kill 
sofia/sip.vonics.net/18582718005 [BREAK]
2007-09-08 01:55:35 [DEBUG] switch_ivr_originate.c:878 
switch_ivr_originate() Originate Cancelled by originator termination 
Cause: 19 [NO_ANSWER]
2007-09-08 01:55:35 [INFO] mod_dptools.c:1087 audio_bridge_function() 
Originate Failed.  Cause: NO_ANSWER

Thanks again for all the help.

Kind Regards,
Chris Danielson

Anthony Minessale wrote:
...
you can find out by playing a file in the script
to make sure you have an audio path.
session.preAnswer();
session.streamFile("/some.wav");
If that doesn't work the inbound device may not support early media so 
try session.answer(); instead to compare what happens when you send a 
200 ok instead of a 183 progress
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
----- Original Message ----
From: Chris Danielson <chris@maxpowersoft.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, September 6, 2007 11:50:29 PM
Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
Thanks again for all the help.  My apologies for not replying sooner, 
I was out of town for a week and needed some extra time to research 
and try some of the things you had mentioned.  Everything that is 
brought up in this thread will get documented by me within the wiki.
So far I have tried both techniques within my JavaScript code that 
Anthony mentioned.  Here is one example:
function cf_bridge_call(phone) {
    var str = "sofia/gateway/line2/"+phone;
    session.setVariable("ringback", "%(2000, 4000, 440.0, 480.0)");
    session.preAnswer();
    session.execute("bridge", str);
}
cf_bridge_call("18881234567");
When I make this call I still do not hear the ringing and notice that 
the console renders:
nta: received 180 Ringing for INVITE (87950735)
nta: 180 Ringing was discarded
When the phone is actually answered, I still cannot hear the person on 
the other side speaking too.
Is this possibly due to the fact that I am having a NAT issue?  Or 
could this be that my carrier is doing something out of the norm?  And 
finally, should I implement this strictly as a dialplan within the 
default_context.xml file and avoid handling this functionality within 
JavaScript?
Kind Regards,
    Chris Danielson
Anthony Minessale wrote:
...
You should choose one or the other not both.
One is inband ringing generated by FreeSWITCH and the other
is just the sip message telling the phone to indicate ringinig.
Here is a description I wrote on an earlier posting here that someone 
may want to stick in the wiki.
---snip
ring_ready is a dialplan application that sends the protocol specific 
message to indicate ringing.  In the case of SIP a 180 Ringing.
ringback is a channel variable you can set to artificially generate a 
tone or play an audio file to an originating channel while it waits 
to be connected to another call.
pre_answer is a dialplan application that will send the protocol 
specific message to indicate early media.  In the case of SIP a 183 
Progress
So, if you call into freeswitch with sip and the first entry in your 
dialplan is ring_ready followed by a bridge to some other destination 
when it hits ring_ready it will send "180 Ringing" back to your phone 
so it can generate the ringing sound.
If instead you use the set application to set ringback to a tone spec 
or audio file followed by a call to pre_answer to establish an early 
media connection followed by a call to bridge to another dest, then 
the core will generate this audio locally and send it back to your 
phone. Again, this occurs during early media meaning the call has not 
been answered but the SIP has negotiated a media path in advance for 
this type of pre-answer audio indication.  SIP has no promise that 
early media must be supported so some switches and devices opt to not 
support it meaning you may not be able to hear any audio until the 
call is officially answered which would keep you from hearing the 
artificial ringback at all.
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
----- Original Message ----
From: Tamas Cseke <cstomi.levlist@gmail.com>
To: chris@maxpowersoft.com; freeswitch-users@lists.freeswitch.org
Sent: Monday, August 27, 2007 2:34:22 AM
Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
Chris Danielson írta:
...
If any one can help me, I am having two issues.  Basically, while my
javascript is executing a single inbound SIP call the script
conditionally gets to a point where the end-user can originate a 
call to
an external phone number.  Keep in mind that I have two sip lines
defined as gateways supplied by my carrier vonics.net.  Upon 
calling the
originate method, the call is actually placed and the destination 
phone
actually begins to ring.  At this point I still cannot hear the 
ringing
on the original session that was originated by my inbound call.  Also,
the originate method times out regardless of whether or not the
destination "phone" number was answered or not.
So my two issues are:
1)  Why does the originate method start a call, the destination phone
called actually rings, but always times out, regardless of whether or
not the phone was answered?
2)  When calling the originate method, I never hear the phone 
ringing on
my inbound connection.
You should call ring_ready app, if you want to hear the ringing afaik.
/* set ringback tone */
                session.setVariable("ringback", "%(2000, 4000, 440.0,
480.0)");
                session.execute("ring_ready", "");
I found these pages, maybe you can find more about it on wiki.
http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
Hope this help!
Regards,
Tamas

Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call

Chris Danielson

Chris Danielson

Chris Danielson

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