[Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
Nick Taylor
nickt at lightlink.com
Thu Sep 13 00:17:26 PDT 2007
I haven't been following this thread and so my answer without reading may
be inappropriate. If so, please kindly disregard.
Port 53 is DNS, it's what's used to translate domain names into ip
addresses. The IP address 192.168.0.1 is a so-called "unroutable"
address, meaning that it can NEVER be a valid address outside of "your own
network". Now, "your own network" could be very large, but if there's no
computer with a 192.168.0.1 address inside of your connection to your
internet service provider, then the line:
> > udp 0 0 10.0.0.2:32826 192.168.0.1:53
> > ESTABLISHED14720/freeswitch
would seem to indicate that DNS is misconfigured somewhere.
If you're on some unix-like machine, please check your /etc/resolv.conf
file.
Anyone more familiar with freeswitch may know that it does or does not use
the system resolver, which I do not know. If it does not use the system
resolver, then the above advice makes no sense whatsoever.
On Wed, 12 Sep 2007, Chris Danielson wrote:
> Date: Wed, 12 Sep 2007 22:44:28 -0700
> From: Chris Danielson <chris at maxpowersoft.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] 2 SIP Lines, Difficulty Originating Call
>
> My apologies for not adding it in my previous message, but my lan
> network is a 10.0.0.x network. The 192.168.0.1 device does not exist on
> my network.
>
> Chris Danielson wrote:
> > After running some more tests I noticed something very interesting...
> > see the 192.168.0.1 line:
> >
> > 2007-09-12 20:49:00 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state()
> > Channel sofia/sip.vonics.net/18881234567 entering state [calling]
> > sres_resolver_receive(0x556cd0, 16) called
> > RR received sip.vonics.net. CNAME IN 890 rdlen=9
> > RR received vonics.net. SOA IN 7199 rdlen=46
> > sres_cache_get(0x549370, SRV, "_sip._udp.sip.vonics.net.") called
> > rr found in cache: _sip._udp.sip.vonics.net. 33
> > rr found in cache: _sip._udp.sip.vonics.net. 33
> > rr found in cache: _sip._udp.sip.vonics.net. 33
> > rr found in cache: _sip._udp.sip.vonics.net. 33
> > sres_cache_get(0x549370, SRV, "_sip._udp.sip.vonics.net.") returned 4
> > entries
> > nta: for "sip.vonics.net" query "_sip._udp.sip.vonics.net" SRV (cached)
> > nta: _sip._udp.sip.vonics.net IN SRV 0 0 5060 sipserver1.vonics.net.
> > (udp)
> > nta: _sip._udp.sip.vonics.net IN SRV 1 0 13412 sipserver1.vonics.net.
> > (udp)
> > nta: _sip._udp.sip.vonics.net IN SRV 2 0 5060 sipserver2.vonics.net.
> > (udp)
> > nta: _sip._udp.sip.vonics.net IN SRV 3 0 13412 sipserver2.vonics.net.
> > (udp)
> > sres_cache_get(0x549370, A, "sipserver1.vonics.net.") called
> > nta: for "sip.vonics.net" query "sipserver1.vonics.net." A
> > sres_query(0x556cd0, 0x690fb0, A, "sipserver1.vonics.net.") called
> > sres_send_dns_query(0x556cd0, 0x691a70) called
> > sres_send_dns_query(0x556cd0, 0x691a70) id=14532 A
> > sipserver1.vonics.net. (to [192.168.0.1]:53)
> > sres_resolver_receive(0x556cd0, 16) called
> > sres_decode_msg: matching query for id=14531
> > sres_resolver_receive(0x556cd0, 16) called
> > RR received sipserver1.vonics.net. CNAME IN 3599 rdlen=9
> > RR received juliet.vonics.net. A IN 4663 rdlen=4
> > RR received vonics.net. NS IN 4663 rdlen=6
> > RR received vonics.net. NS IN 4663 rdlen=6
> > RR received vonics.net. NS IN 4663 rdlen=6
> >
> > And following this log I ran a netstat just to see the socket
> >
> > MaxPowerSoft001:~# netstat -nalp |grep freeswitch
> > tcp 0 0 10.0.0.2:13412 0.0.0.0:*
> > LISTEN 14720/freeswitch
> > udp 0 0 10.0.0.2:32826 192.168.0.1:53
> > ESTABLISHED14720/freeswitch
> > udp 0 0 10.0.0.2:13412
> > 0.0.0.0:* 14720/freeswitch
> >
> >
> > I am wondering if the 192.168.0.1 address is causing my issues.
> > Thanks again for any light you shine on this issue.
> >
> > Kind Regards,
> > Chris Danielson
> >
> > Chris Danielson wrote:
> >> 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 at 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/18881234567 [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 at hotmail.com
> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>> IRC: irc.freenode.net #freeswitch
> >>>
> >>> FreeSWITCH Developer Conference
> >>> sip:888 at conference.freeswitch.org
> >>> iax:guest at conference.freeswitch.org/888
> >>> googletalk:conf+888 at conference.freeswitch.org
> >>> pstn:213-799-1400
> >>>
> >>>
> >>> ----- Original Message ----
> >>> From: Chris Danielson <chris at maxpowersoft.com>
> >>> To: freeswitch-users at 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 at hotmail.com
> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>>> IRC: irc.freenode.net #freeswitch
> >>>>
> >>>> FreeSWITCH Developer Conference
> >>>> sip:888 at conference.freeswitch.org
> >>>> iax:guest at conference.freeswitch.org/888
> >>>> googletalk:conf+888 at conference.freeswitch.org
> >>>> pstn:213-799-1400
> >>>>
> >>>>
> >>>> ----- Original Message ----
> >>>> From: Tamas Cseke <cstomi.levlist at gmail.com>
> >>>> To: chris at maxpowersoft.com; freeswitch-users at 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
> >
>
>
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