Thanks Anthony, it's here: http://pastebin.freeswitch.org/14350 And pcap is attached. The call connects around (or just before) the 16:58:35 mark (line 558 is what I see in the terminal while waiting for it to connect - both early-media and the missing start of the media) Cheers, Fraser 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${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D> "/> <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?
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-- Anthony Minessale II
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AIM: anthm MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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