OK so, The phone sends a 180 ringing with NO SDP then it starts sending RTP That's is not right. It's a bug in the phone. On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you try another one with just udp and not "port 5060" so I can see the rtp too
On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
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${destination_number}"/> > > <action application="bridge" data="${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 > > _______________________________________________ > 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
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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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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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
_______________________________________________ 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
_______________________________________________ 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
-- 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