On Feb 28, 2011 5:20 PM, "Anthony Minessale" <anthony.minessale <at> gmail.com> wrote:
> You did not mention that you are originating the call over event
> socket or some script.
> Check your sofia profile xml and make sure you have these set.
>
> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>
> You are offering every single codec ever at the same time, many of
> them will not go well with each other in the same offer.
>
>
>
> On Mon, Feb 28, 2011 at 5:12 PM, Malay Thakershi <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>> I am including everything. First line - until the end.
>>
>> On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale
>> <anthony.minessale <at> gmail.com> wrote:
>>>
>>> You are not including the entire log going back to when the call first
>>> HITS FS.
>>> We need to see the initial invite in the log.
>>>
>>>
>>>
>>> On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
>>> wrote:
>>> > http://pastebin.freeswitch.org/15500
>>> > Thank you.
>>> > Malay
>>> >
>>> > On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale
>>> > <anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>> >>
>>> >> If you read the dialog box closely the login and password are stated
>>> >> in the challenge box.
>>> >> That is our way of telling who pays attention and who doesn't...
>>> >>
>>> >> not really, its to keep spambots off the site.
>>> >>
>>> >>
>>> >> On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
>>> >> wrote:
>>> >> > I can't login to FS pastebin.
>>> >> > I used same user / password as I use to access users group.
>>> >> > Malay
>>> >> >
>>> >> > On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
>>> >> > <anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>> >> >>
>>> >> >> You also need "sofia global siptrace on" so we can see the SIP
>>> >> >> traffic
>>> >> >> in the log.
>>> >> >> We have our own pastebin at http://pastebin.freeswitch.org
>>> >> >>
>>> >> >>
>>> >> >> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi
>>> >> >> <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
>>> >> >> wrote:
>>> >> >> > http://pastebin.com/CCwmAfZK
>>> >> >> > I noticed that LPC warning only comes for outbound calls I
>>> >> >> > initiate
>>> >> >> > using
>>> >> >> > "Originate" API.
>>> >> >> > Malay
>>> >> >> >
>>> >> >> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>>> >> >> > <anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>> >> >> >>
>>> >> >> >> That's L16 not LPC, you need L16 to play files.
>>> >> >> >> Why don't you just put the whole log of the call on pastebin.
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi
>>> >> >> >> <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
>>> >> >> >> wrote:
>>> >> >> >> > Following is a section from the log:
>>> >> >> >> > ---------------------------
>>> >> >> >> > 2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec
>>> >> >> >> > sofia/sipinterface_1/########## <at> ##.##.##.## PCMU/8000 20 ms 160
>>> >> >> >> > samples
>>> >> >> >> > 64000 bits
>>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298
>>> >> >> >> > OPEN
>>> >> >> >> > TTS
>>> >> >> >> > cepstral
>>> >> >> >> > 2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307
>>> >> >> >> > Raw
>>> >> >> >> > Codec
>>> >> >> >> > Activated
>>> >> >> >> > 2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996
>>> >> >> >> > Speaking
>>> >> >> >> > text:
>>> >> >> >> > <break strength='medium'/>We must verify your identity.
>>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188
>>> >> >> >> > done
>>> >> >> >> > speaking
>>> >> >> >> > text
>>> >> >> >> > 2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244
>>> >> >> >> > Codec
>>> >> >> >> > Activated L16 <at> 8000hz 1 channels 20ms
>>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581
>>> >> >> >> > done
>>> >> >> >> > playing
>>> >> >> >> > file
>>> >> >> >> > 2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec
>>> >> >> >> > Activated
>>> >> >> >> > L16 <at> 8000hz 1 channels 20ms
>>> >> >> >> > 2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244
>>> >> >> >> > Codec
>>> >> >> >> > Activated L16 <at> 8000hz 1 channels 20ms
>>> >> >> >> > ---------------------------
>>> >> >> >> > So I see when I play a file (using StreamFile / PAGD), it
>>> >> >> >> > activates
>>> >> >> >> > L16,
>>> >> >> >> > which the wiki pages says is not recommended. So should I
>>> >> >> >> > deactivate
>>> >> >> >> > it?
>>> >> >> >> > If
>>> >> >> >> > so, how?
>>> >> >> >> > Now, I have not done any setting out of default / ordinary that
>>> >> >> >> > comes
>>> >> >> >> > with
>>> >> >> >> > the build. I am playing WAV file that is generated by Cepstral
>>> >> >> >> > SWIFT
>>> >> >> >> > command
>>> >> >> >> > line tool (text to WAV). The file format is "Wave PCM signed 16
>>> >> >> >> > bit,
>>> >> >> >> > 8000
>>> >> >> >> > Hz, 128 kbps, mono".
>>> >> >> >> > Thank you for help so far.
>>> >> >> >> > Malay
>>> >> >> >> > On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale
>>> >> >> >> > <anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>> >> >> >> >>
>>> >> >> >> >> look at your SIP traffic and console log.
>>> >> >> >> >>
>>> >> >> >> >> enter "sofia global siptrace on" followed by "console loglevel
>>> >> >> >> >> debug"
>>> >> >> >> >> at the cli and make the call.
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi
>>> >> >> >> >> <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org>
>>> >> >> >> >> wrote:
>>> >> >> >> >> > I have no idea where to look for this setting.
>>> >> >> >> >> > This is in modules.conf.xml
>>> >> >> >> >> >     <!-- Codec Interfaces -->
>>> >> >> >> >> >     <load module="mod_spandsp"/>
>>> >> >> >> >> >     <!--<load module="mod_voipcodecs"/>-->
>>> >> >> >> >> >     <load module="mod_g723_1"/>
>>> >> >> >> >> >     <load module="mod_g729"/>
>>> >> >> >> >> >     <load module="mod_amr"/>
>>> >> >> >> >> >     <load module="mod_ilbc"/>
>>> >> >> >> >> >     <load module="mod_speex"/>
>>> >> >> >> >> >     <load module="mod_h26x"/>
>>> >> >> >> >> >     <load module="mod_siren"/>
>>> >> >> >> >> >     <!--<load module="mod_celt"/>-->
>>> >> >> >> >> >     <!--<load module="mod_opus"/>-->
>>> >> >> >> >> > Apart from settings I posted in my previous post, where else
>>> >> >> >> >> > to
>>> >> >> >> >> > look
>>> >> >> >> >> > for
>>> >> >> >> >> > disabling LPC?
>>> >> >> >> >> > Malay
>>> >> >> >> >> > On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
>>> >> >> >> >> > <anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>> >> >> >> >> >>
>>> >> >> >> >> >> is your inbound call using LPC? you don't want to be using
>>> >> >> >> >> >> LPC
>>> >> >> >> >> >> and
>>> >> >> >> >> >> expect anything to sound good that's for sure.
>>> >> >> >> >> >> It would not just magically say that unless something you
>>> >> >> >> >> >> are
>>> >> >> >> >> >> doing
>>> >> >> >> >> >> has
>>> >> >> >> >> >> LPC?
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi
>>> >> >> >> >> >> <mthakershi <at> gmail.com>
>>> >> >> >> >> >> wrote:
>>> >> >> >> >> >> > Hello,
>>> >> >> >> >> >> > I updated to the latest FS version last week.
>>> >> >> >> >> >> > I started getting the following warning when speech /
>>> >> >> >> >> >> > sound
>>> >> >> >> >> >> > is
>>> >> >> >> >> >> > played
>>> >> >> >> >> >> > on
>>> >> >> >> >> >> > the
>>> >> >> >> >> >> > call.
>>> >> >> >> >> >> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213
>>> >> >> >> >> >> > Codec
>>> >> >> >> >> >> > LPC
>>> >> >> >> >> >> > payload
>>> >> >> >> >> >> > 7
>>> >> >> >> >> >> > added to sdp wanting ptime 90 but it's already 20
>>> >> >> >> >> >> > (G7221:115:20),
>>> >> >> >> >> >> > disabling
>>> >> >> >> >> >> > ptime."
>>> >> >> >> >> >> > I read sections on codecs and negotiations.
>>> >> >> >> >> >> > Following are the settings from vars.xml (I have not
>>> >> >> >> >> >> > changed
>>> >> >> >> >> >> > them):
>>> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> > data="global_codec_prefs=G7221 <at> 32000h,G7221 <at> 16000h,G722,PCMU,PCMA,GSM"/>
>>> >> >> >> >> >> >   <X-PRE-PROCESS cmd="set"
>>> >> >> >> >> >> > data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>> >> >> >> >> >> > Also, there is no codec related setting in sip_profiles
>>> >> >> >> >> >> > files
>>> >> >> >> >> >> > and sofia.conf.xml file.
>>> >> >> >> >> >> > I am playing audio files using Cepstral TTS during the
>>> >> >> >> >> >> > call.
>>> >> >> >> >> >> > Can someone please help me understand these settings? And
>>> >> >> >> >> >> > if
>>> >> >> >> >> >> > they
>>> >> >> >> >> >> > are
>>> >> >> >> >> >> > appropriate?
>>> >> >> >> >> >> > Thank you.
>>> >> >> >> >> >> > Malay
>>> >> >> >> >> >> > _______________________________________________
>>> >> >> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> >> >> > FreeSWITCH-users-PD4FTy7X32nootDcgO//1Q@public.gmane.orgtch.org
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> >> > http://www.freeswitch.org
>>> >> >> >> >> >> >
>>> >> >> >> >> >> >
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >> --
>>> >> >> >> >> >> Anthony Minessale II
>>> >> >> >> >> >>
>>> >> >> >> >> >> FreeSWITCH http://www.freeswitch.org/
>>> >> >> >> >> >> ClueCon http://www.cluecon.com/
>>> >> >> >> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >> >> >> >>
>>> >> >> >> >> >> AIM: anthm
>>> >> >> >> >> >> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> >> >> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale <at> gmail.com
>>> >> >> >> >> >> IRC: irc.freenode.net #freeswitch
>>> >> >> >> >> >>
>>> >> >> >> >> >> FreeSWITCH Developer Conference
>>> >> >> >> >> >> sip:888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> >> >> >> >> googletalk:conf+888 <at> conference.freeswitch.org
>>> >> >> >> >> >> pstn:+19193869900
>>> >> >> >> >> >>
>>> >> >> >> >> >> _______________________________________________
>>> >> >> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> >> >> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW+46SIhDJjyZ@public.gmane.orgrg
>>> >> >> >> >> >>
>>> >> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >>
>>> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> >> http://www.freeswitch.org
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> > _______________________________________________
>>> >> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> >> > FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> >> >> >
>>> >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> > http://www.freeswitch.org
>>> >> >> >> >> >
>>> >> >> >> >> >
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> --
>>> >> >> >> >> Anthony Minessale II
>>> >> >> >> >>
>>> >> >> >> >> FreeSWITCH http://www.freeswitch.org/
>>> >> >> >> >> ClueCon http://www.cluecon.com/
>>> >> >> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >> >> >>
>>> >> >> >> >> AIM: anthm
>>> >> >> >> >> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> >> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale <at> gmail.com
>>> >> >> >> >> IRC: irc.freenode.net #freeswitch
>>> >> >> >> >>
>>> >> >> >> >> FreeSWITCH Developer Conference
>>> >> >> >> >> sip:888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> >> >> >> googletalk:conf+888 <at> conference.freeswitch.org
>>> >> >> >> >> pstn:+19193869900
>>> >> >> >> >>
>>> >> >> >> >> _______________________________________________
>>> >> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> >> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> http://www.freeswitch.org
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > _______________________________________________
>>> >> >> >> > FreeSWITCH-users mailing list
>>> >> >> >> > FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> > http://www.freeswitch.org
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> --
>>> >> >> >> Anthony Minessale II
>>> >> >> >>
>>> >> >> >> FreeSWITCH http://www.freeswitch.org/
>>> >> >> >> ClueCon http://www.cluecon.com/
>>> >> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >> >>
>>> >> >> >> AIM: anthm
>>> >> >> >> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org
>>> >> >> >> IRC: irc.freenode.net #freeswitch
>>> >> >> >>
>>> >> >> >> FreeSWITCH Developer Conference
>>> >> >> >> sip:888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> >> >> googletalk:conf+888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> >> >> pstn:+19193869900
>>> >> >> >>
>>> >> >> >> _______________________________________________
>>> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> http://www.freeswitch.org
>>> >> >> >
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >
>>> >> >> >
>>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> > http://www.freeswitch.org
>>> >> >> >
>>> >> >> >
>>> >> >>
>>> >> >>
>>> >> >>
>>> >> >> --
>>> >> >> Anthony Minessale II
>>> >> >>
>>> >> >> FreeSWITCH http://www.freeswitch.org/
>>> >> >> ClueCon http://www.cluecon.com/
>>> >> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >> >>
>>> >> >> AIM: anthm
>>> >> >> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org
>>> >> >> IRC: irc.freenode.net #freeswitch
>>> >> >>
>>> >> >> FreeSWITCH Developer Conference
>>> >> >> sip:888 <at> conference.freeswitch.org
>>> >> >> googletalk:conf+888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> >> pstn:+19193869900
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >>
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users <at> lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Anthony Minessale II
>>> >>
>>> >> FreeSWITCH http://www.freeswitch.org/
>>> >> ClueCon http://www.cluecon.com/
>>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >> AIM: anthm
>>> >> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> googletalk:conf+888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> >> pstn:+19193869900
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users-PD4FTy7X32k@public.gmane.orgeeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW2D2FQJk+8+b@public.gmane.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org
>>> GTALK/JABBER/PAYPAL:anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> googletalk:conf+888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRW+G/Ez6ZCGd0@public.gmane.orgorg
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users-PD4FTy7X32k3T7AR8gHRWw@public.gmane.org.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale <at> hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 <at> conference.freeswitch.org
> googletalk:conf+888-8PHf0o9tX6c3pKE/1MXUkG7psfBIVZK/2LY78lusg7I@public.gmane.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
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On Mon, Feb 28, 2011 at 4:34 PM, Malay Thakershi <mthakershi-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:

Ok.
I am using managed esl to send originate.

I will put these params in sofia profiles.

You said I am offering all codecs at once. How do I restrict this? And what should be the codec offered in my case?

On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman <freeswitch-jvZMY70mQyGWw5T4fKrT70EOCMrvLtNR@public.gmane.org> wrote:

What FS build are you running?  I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600).  What about inbound?  Do you have that set up?

Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com

On 02/28/2011 12:54 AM, envelopes envelopes wrote:

never mind. add this line <action application="ring_ready"/> and restart FS

fixed the issue.

On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <sunwood360-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:

Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call.
is there any config variable to set up so that I will be notified if my call is answered?

thanks!

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Can you reproduce the sound problems on the very latest Git?

Can you
hear speech with noise, or is it just garbage?

Can you get a
debug-level log for the call with siptrace enabled so we can see
what's going on? Also use tcpdump/wireshark/tshark to capture the RTP
and analyze it in Wireshark to check for packet loss or jitter.

On Mar 1, 2011 4:59 AM, "envelopes envelopes" <sunwood360-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
> I don't use GV for direct inbound, because it is not working very well.
>
> - Get an account from sipgate.com;
> - let GV forward inbound to your sipgate PSTN #
> - and your FS register as a sipgate client.
>
>
>
> On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman <
> freeswitch <at> cartissolutions.com> wrote:
>
>> What FS build are you running? I'd like to compare notes with you on your
>> config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version
>> 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do
>> you have that set up?
>>
>> Yossi Neiman
>> Cartis Solutions, Inc. - http://www.cartissolutions.com
>>
>> On 02/28/2011 12:54 AM, envelopes envelopes wrote:
>>
>> never mind. add this line <action application="ring_ready"/> and restart FS
>>
>> fixed the issue.
>>
>>
>>
>> On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <
>> sunwood360-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>>
>>> Now I am able to use GV for outbound dialing. However, I don't hear any
>>> ringback or not sure whether the other party has answered the call.
>>> is there any config variable to set up so that I will be notified if my
>>> call is answered?
>>>
>>> thanks!
>>>
>>>
>>
>> _______________________________________________
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