On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Can you do this trace with debug level logging in addition to the sip trace
console loglevel debug

you also may want to get a pcap of it

tshark udp and port 5060 -w test.pcap

On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond

<fraserredmond@gmail.com> wrote:
> Thanks Anthony,
>
> Finally managed to get a sip trace - could you do me a favor and take a look
> and/or give me some ideas of what to look for?
>
> http://pastebin.freeswitch.org/14300
>
> I've highlighted lines 168 and 193. In between these lines is where the
> number is dialed and rings once, then picks up, then theres silence for a
> second or two, and that second SIP message is when I start hearing audio.
>
> Thanks,
> Fraser
>
>
>
>
> On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale
> <anthony.minessale@gmail.com> wrote:
>>
>> its a blue message on cli
>>
>> It could also be the other side expecting us to send media first or
>> something silly.
>> try getting a sip trace and figure out when the rtp starts arriving.
>>
>>
>> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
>> <fraserredmond@gmail.com> wrote:
>> > Sorry, yes, I am setting ignore_early_media=true in the first area. (Or
>> > are
>> > you saying that should be off? I forget now why I needed it on, but
>> > there
>> > was a reason I added it.)
>> >
>> > Yes, the bridge doesn't start until after the A-leg has answered.
>> >
>> > Thanks for the suggestion about nat/auto-changing port, I'll have a look
>> > into that - would that be in the cli output or in a sip trace? I've
>> > already
>> > looked and it's not appearing in the CLI output (with loglevel=debug),
>> > haven't looked in the sip trace yet.
>> >
>> > Cheers,
>> > Fraser
>> >
>> >
>> >
>> >
>> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
>> > <anthony.minessale@gmail.com> wrote:
>> >>
>> >> are you setting ignore_early_media=true in the first vars=values area?
>> >>
>> >> This looks like you could be calling one leg who is still not answered
>> >> and then bridging it to another dest. The bridge app will wait for
>> >> the first leg to answer before bridging.
>> >>
>> >> Also if you have any NAT anywhere, look for an "auto-changing port"
>> >> type message which can also be attributed to this due to a detection
>> >> period for incorrect ports.
>> >>
>> >>
>> >>
>> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
>> >> <fraserredmond@gmail.com> wrote:
>> >> > event_socket:
>> >> > api originate {vars=values}user/$fromExtn@Domain
>> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
>> >> >
>> >> > then
>> >> >
>> >> > dialplan:
>> >> > <action application="set"
>> >> > data="effective_caller_id_number=+1800number"/>
>> >> > <action application="set" data="effective_caller_id_name="/>
>> >> > (set and/or export a bunch of other vars too)
>> >> > <action application="set"
>> >> >
>> >> >
>> >> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}"/>
>> >> > <action application="bridge" data="${dial_string}"/>
>> >> >
>> >> > Cheers,
>> >> > Fraser
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
>> >> > <anthony.minessale@gmail.com> wrote:
>> >> >>
>> >> >> how are you accomplishing that? by which technique?
>> >> >>
>> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> >> >> <fraserredmond@gmail.com> wrote:
>> >> >> > The call is originated from Freeswitch (via CLI) to a softphone,
>> >> >> > then
>> >> >> > when
>> >> >> > that is connected it bridges out to the gateway.
>> >> >> >
>> >> >> > Cheers,
>> >> >> > Fraser
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
>> >> >> > <anthony.minessale@gmail.com> wrote:
>> >> >> >>
>> >> >> >> Where is the other side of this call coming from?
>> >> >> >>
>> >> >> >> [ ( ) ] -> FS -> (PSTN via SIP)
>> >> >> >>
>> >> >> >> What goes in the empty space above?
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users@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@hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888@conference.freeswitch.org
>> >> googletalk:conf+888@conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
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>> >>
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>> >
>> >
>> > _______________________________________________
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>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> 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
>> googletalk:conf+888@conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
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>
>
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>
>

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

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