please set the environment variable TPORT_LOG=1 to log the sip traffic inline. then capture all of the console output reproduce your issue before you try the call, please enter the command "console loglevel debug" at the CLI also, just to be safe, examine your dialplan for any instances of bypass_media and examine all of your sip profiles for the inbound-bypass-media param. On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Thank you very much, Anthony, for your answer!
Anthony Minessale napsal(a):
do you have bypass_media=true set or the inbound-bypass-media sip profile option set?
No, no. I use neither of them (if they are not default :-).
BTW: I need FreeSWITCH in media path due to four reasons: 1. To handle NATed clients. 2. To kill call without RTP. 2. To process DTMF. 3. To transcode between codecs if needed.
This call flow looks to me like proxy mode where the sdp of the inbound leg is offered to the remote leg to attempt to establish direct communication between caller and gateway.
If yes it is (my?) mistake. I don't want proxy mode :-)
Since caller has natted IP this may not work so well.
I think so.
If you are setting bypass_media=true, either remove it or use proxy_media=true instead.
May be - in the future - i will utilize proxy_media=true too, but now I need two independant call legs with two independant media streams with FreeSWITCH between them.
If you have any suggestion what should I do to eliminate my trouble, please give me the clue. I'm afraid I have no idea what else I could try to do :-)
Thanks in advance, best regards
kokoska.rokoska
On Sun, Apr 20, 2008 at 10:39 AM, kokoska rokoska <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
My be i have a clue :-)
May be FreeSWITCH thinks that offers different IPs because I set "ext-rtp-ip" to public IP of my network in SIP profile gateway is in
FreeSWITCH runs on devel machine behind NAT...
Do you think it could be the reason? And if yes, how to avoid it?
Best regards,
kokoska.rokoska
kokoska rokoska napsal(a): > > Brian West napsal(a): >> Can you provide a bit more detail? I can only guess what or how >> you're trying to use it. >> > > Well, here is what I try to do: > > 1. I have two users defined in directory which are registered
with
> FreeSWITCH. Both are in same SIP profile "default" (FS UA runs at port > 5065). > > 2. I have different SIP profile (FS UA runs at port 7001) in
which I
> have defiend gateway to my telco-provider. > > 3. A make a call from one of users (see 1.) to some PSTN number. > FreeSWITCH (UA at port 5065) receives INVITE, ask about credentials etc. > FreeSWITCH (UA at port 7001) successfuly authorizes against my
telco
> etc. and establish media flow (183,200) from local port, say
1234, IP
> 1.2.3.4 <http://1.2.3.4>. > > 4. Than UA at port 5065 try to establish media flow (183,200) to local > user, but offers the same IP/port in SDP - 1.2.3.4:1234 <http://1.2.3.4:1234>. > > 5. Than both legs (my telco and local user) send RTP to same
IP:port.
> IMO it can't work. And, like I think, it don't - both legs hear nothing > and FreeSWITCH kills call after a while with cause:
MEDIA_TIMEOUT.
> > >> How are you trying to call a registered user? > > I'm not sure what you are asking. I call localy registred user by > <action application='bridge' data='sofia/default/user%domain'/> > and it works fine. > > What don't work (see above) is to call out through gateway: > <action application='bridge'
data='sofia/gateway/gw_name/number'/>
> >> It's perfectly OK to >> receive media from two different IP's on the same port > > Yes, but there is same IP and port... > >> but I smell a >> bug but more so a usage case we haven't tested. >> > > Not sure, but it looks like it. Or I miss something important in config > options. > >> Provide the sip trace and the console log on pastebin and reply
with
>> the urls. >> > > Here is pastebin url: > http://pastebin.freeswitch.org/4273 > > > Best regards, > > kokoska.rokoska > >> /b >> >> On Apr 20, 2008, at 5:42 AM, kokoska rokoska wrote: >> >>> Hi all, >>> >>> I have came into troubles with calling from localy registered
user
>>> through Sofia gateway: >>> >>> FreeSWITCH offers the same RTP port in SDP to both call legs and thus >>> I'm without audio and FreeSWITCH kills the call after a while
with
>>> cause >>> MEDIA_TIMEOUT (this is not really valid cause, because media goes to >>> FreeSWITCH from both legs, but to the same port :-) >>> >>> When I make call between localy registered users, everything
looks
>>> good. >>> >>> I really don't know what I'm doing wrong, so any clue is very >>> appreciated :-) >>> >>> BTW: I have both FreeSWITCH console log and pcap dump if
someone
>>> interested. And, of course, can supply any other required informations >>> about calls. >>> >>> Thanks in advance, best regards >>> >>> kokoska.rokoska >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users@lists.freeswitch.org <mailto: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 >> Brian West >> sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org> <mailto:
sip%3Abrian@freeswitch.org <sip%253Abrian@freeswitch.org>>
>> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org <mailto: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 <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400