Hi! Hopefully an easy question... What is the parameter to define the authorization username for a gateway? <param name="username" value="didnumber"/> <param name="password" value="password"/> I also need something like: <param name="auth-user" value="username"/> I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work.... Thanks! /Mark
Try from-user and from-domain /b On Apr 19, 2008, at 5:30 PM, Mark Campbell-Smith wrote:
Hi!
Hopefully an easy question...
What is the parameter to define the authorization username for a gateway?
<param name="username" value="didnumber"/> <param name="password" value="password"/>
I also need something like: <param name="auth-user" value="username"/>
I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work....
Thanks! /Mark
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Brian West sip:brian@freeswitch.org
btw the username is the used in the authorization header.. what isn't working for you? Can you elaborate on what isn't working or email me off list your gateway details so I can compose you the xml required? Then maybe you can document this on the wiki? /b On Apr 19, 2008, at 5:30 PM, Mark Campbell-Smith wrote:
Hi!
Hopefully an easy question...
What is the parameter to define the authorization username for a gateway?
<param name="username" value="didnumber"/> <param name="password" value="password"/>
I also need something like: <param name="auth-user" value="username"/>
I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work....
Thanks! /Mark
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Brian West sip:brian@freeswitch.org
Here are all the possible values for a gateway: (someone can wikify this) Mark if none of this clears up send me the account details and i'll whip up some xml. register (true or false like when you don't want to register but want to auth to a gateway) schema (Currently should be left blank which is Digest by default) realm (authorization realm in some cases you might need to set this) username (this is the authorization and from username in the registration unless from-user is set.) password (well the password) caller-id-in-from (for those times the callerid is in the from for when the remote doesn't do rpid or p-asserted) extensions (this is the contact extension for the remote to contact) proxy (defaults to the gateway name unless you set this param) context (what context this gateway is in) expire-seconds (how many seconds to register for default is 3600 if left empty) retry-seconds (this sets the number of seconds to retry the registration) from-user (the username used in the from field defaults to the username if left empty) from-domain (the domain to use in the from field) register-proxy (for those times you have a proxy you register with and a different proxy to send calls to defaults to gateway name or proxy if left empty) contact-params (extra contact params ie tport=tcp) register-transport (udp,tcp or tls if you have it enabled) Hope this helps someone. /b On Apr 19, 2008, at 5:30 PM, Mark Campbell-Smith wrote:
Hi!
Hopefully an easy question...
What is the parameter to define the authorization username for a gateway?
<param name="username" value="didnumber"/> <param name="password" value="password"/>
I also need something like: <param name="auth-user" value="username"/>
I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work....
Thanks! /Mark
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http://wiki.freeswitch.org/wiki/Authentication -wasim On Sun, Apr 20, 2008 at 7:12 AM, Brian West <brian@freeswitch.org> wrote:
Here are all the possible values for a gateway: (someone can wikify this)
Mark if none of this clears up send me the account details and i'll whip up some xml.
register (true or false like when you don't want to register but want to auth to a gateway)
schema (Currently should be left blank which is Digest by default)
realm (authorization realm in some cases you might need to set this)
username (this is the authorization and from username in the registration unless from-user is set.)
password (well the password)
caller-id-in-from (for those times the callerid is in the from for when the remote doesn't do rpid or p-asserted)
extensions (this is the contact extension for the remote to contact)
proxy (defaults to the gateway name unless you set this param)
context (what context this gateway is in)
expire-seconds (how many seconds to register for default is 3600 if left empty)
retry-seconds (this sets the number of seconds to retry the registration)
from-user (the username used in the from field defaults to the username if left empty)
from-domain (the domain to use in the from field)
register-proxy (for those times you have a proxy you register with and a different proxy to send calls to defaults to gateway name or proxy if left empty)
contact-params (extra contact params ie tport=tcp)
register-transport (udp,tcp or tls if you have it enabled)
Hope this helps someone.
/b
On Apr 19, 2008, at 5:30 PM, Mark Campbell-Smith wrote:
Hi!
Hopefully an easy question...
What is the parameter to define the authorization username for a gateway?
<param name="username" value="didnumber"/> <param name="password" value="password"/>
I also need something like: <param name="auth-user" value="username"/>
I've tried auth-user, auth-username, extra-auth-user etc, but none seem to work....
Thanks! /Mark
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-- wasim h. baig | principal consultant | convergence pk | +92 300 8508070
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
Can you provide a bit more detail? I can only guess what or how you're trying to use it. How are you trying to call a registered user? It's perfectly OK to receive media from two different IP's on the same port but I smell a bug but more so a usage case we haven't tested. Provide the sip trace and the console log on pastebin and reply with the urls. /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
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Brian West sip:brian@freeswitch.org
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. 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. 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
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Brian West sip:brian@freeswitch.org
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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.
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.
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 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
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do you have bypass_media=true set or the inbound-bypass-media sip profile option set? 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. Since caller has natted IP this may not work so well. If you are setting bypass_media=true, either remove it or use proxy_media=true instead. On Sun, Apr 20, 2008 at 10:39 AM, kokoska rokoska <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.
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.
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
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http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org Brian West sip:brian@freeswitch.org <sip%3Abrian@freeswitch.org>
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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
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 <mailto:sip%3Abrian@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: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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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:
http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ 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
-- 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>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400
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_______________________________________________ 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/ 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
Anthony Minessale napsal(a):
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.
Many thanks, Anthony, for your help! I double-checked my sip profiles and dialplan and found nothing which looks like "bypass"... Console log I have submited to pastebin: http://pastebin.freeswitch.org/4277 Best regards, kokoska.rokoska
On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska <kokoska.rokoska@post.cz <mailto: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> <mailto: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> <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> > <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> > <mailto: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 <mailto:sip%3Abrian@freeswitch.org> <mailto:sip%3Abrian@freeswitch.org <mailto:sip%253Abrian@freeswitch.org>> > >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto: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> > <mailto: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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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Please do the same trace but actually make the phone call that you were describing in your original email so i can see the output of the FS console and the SIP trace together. Also see line 38 of your pastebin there appear to be some xml errs in the data you are returning from xml-curl On Mon, Apr 21, 2008 at 10:43 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
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.
Many thanks, Anthony, for your help!
I double-checked my sip profiles and dialplan and found nothing which looks like "bypass"...
Console log I have submited to pastebin:
http://pastebin.freeswitch.org/4277
Best regards,
kokoska.rokoska
On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska <kokoska.rokoska@post.cz <mailto: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> <mailto: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> <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> > <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> > <mailto: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>> <mailto:sip%3Abrian@freeswitch.org <sip%253Abrian@freeswitch.org> <mailto:sip%253Abrian@freeswitch.org<sip%25253Abrian@freeswitch.org>
> >> > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto: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
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> http://www.freeswitch.org > > > > > -- > 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>
> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
> pstn:213-799-1400 > > >
------------------------------------------------------------------------
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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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IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
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pstn:213-799-1400
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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
Anthony Minessale napsal(a):
Please do the same trace but actually make the phone call that you were describing in your original email so i can see the output of the FS console and the SIP trace together.
I made the same call I decribed in my previous e-mail, but now I see pastebin accepts only few lines (about 600) - not all of them, thou you can't see mentioned call debug :-) I grab new FreeSWITCH debug and attached it to this e-mail, beacuse I didn't find the way how to copy full debug to pastebin...
Also see line 38 of your pastebin there appear to be some xml errs in the data you are returning from xml-curl
Thank you very much, Anthony, for that point! This happens on Directory post without "name" and "ip". I don't know what is this post for (nothing found in wiki) and forget to reply with something relevant. Now I reply with 404. BTW: Could you (or anybody else) tell me what answer FreeSWITCH expects to the post above? It is on line 35 in debug... Best regards, kokoska.rokoska
On Mon, Apr 21, 2008 at 10:43 AM, kokoska rokoska <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
Anthony Minessale napsal(a): > 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.
Many thanks, Anthony, for your help!
I double-checked my sip profiles and dialplan and found nothing which looks like "bypass"...
Console log I have submited to pastebin:
http://pastebin.freeswitch.org/4277
Best regards,
kokoska.rokoska
> > > On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska > <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz> <mailto:kokoska.rokoska@post.cz <mailto: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> <mailto:kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> > <mailto:kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz> <mailto: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> <http://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> > <http://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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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 <mailto:sip%3Abrian@freeswitch.org> > <mailto:sip%3Abrian@freeswitch.org <mailto:sip%253Abrian@freeswitch.org>> > <mailto:sip%3Abrian@freeswitch.org <mailto:sip%253Abrian@freeswitch.org> > <mailto:sip%253Abrian@freeswitch.org <mailto:sip%25253Abrian@freeswitch.org>>> > > >> > > >> > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com> > <mailto:MSN%253Aanthony_minessale@hotmail.com <mailto:MSN%25253Aanthony_minessale@hotmail.com>>> > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com> > <mailto:PAYPAL%253Aanthony.minessale@gmail.com <mailto:PAYPAL%25253Aanthony.minessale@gmail.com>>> > > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> > <http://irc.freenode.net> #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org> > <mailto:sip%253A888@conference.freeswitch.org <mailto:sip%25253A888@conference.freeswitch.org>>> > > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > > <http://iax:guest@conference.freeswitch.org/888> > > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org> > <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org <mailto:googletalk%25253Aconf%25252B888@conference.freeswitch.org>>> > > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto: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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony_minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------------------------------------------------
_______________________________________________ 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
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414codec: 8 ms: 20 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065] 192.168.1.128:21322 ->192.168.1.101\ :49152 codec: 8 ms: 20 Are you setting something in your config to use 0.0.0.0 as the media IP? Perhaps putting the real ip in ext-rtp-ip and setting the local one to 0.0.0.0 ? On Mon, Apr 21, 2008 at 12:08 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
Please do the same trace but actually make the phone call that you were describing in your original email so i can see the output of the FS console and the SIP trace together.
I made the same call I decribed in my previous e-mail, but now I see pastebin accepts only few lines (about 600) - not all of them, thou you can't see mentioned call debug :-)
I grab new FreeSWITCH debug and attached it to this e-mail, beacuse I didn't find the way how to copy full debug to pastebin...
Also see line 38 of your pastebin there appear to be some xml errs in the data you are returning from xml-curl
Thank you very much, Anthony, for that point! This happens on Directory post without "name" and "ip". I don't know what is this post for (nothing found in wiki) and forget to reply with something relevant. Now I reply with 404.
BTW: Could you (or anybody else) tell me what answer FreeSWITCH expects to the post above? It is on line 35 in debug...
Best regards,
kokoska.rokoska
On Mon, Apr 21, 2008 at 10:43 AM, kokoska rokoska < kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
Anthony Minessale napsal(a): > 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.
Many thanks, Anthony, for your help!
I double-checked my sip profiles and dialplan and found nothing which looks like "bypass"...
Console log I have submited to pastebin:
http://pastebin.freeswitch.org/4277
Best regards,
kokoska.rokoska
> > > On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska > <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz> <mailto:kokoska.rokoska@post.cz <mailto: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> <mailto:kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> > <mailto:kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz> <mailto: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> <http://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> > <http://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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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>> > <mailto:sip%3Abrian@freeswitch.org<sip%253Abrian@freeswitch.org> <mailto:sip%253Abrian@freeswitch.org <sip%25253Abrian@freeswitch.org>
> <mailto:sip%3Abrian@freeswitch.org<sip%253Abrian@freeswitch.org> <mailto:sip%253Abrian@freeswitch.org <sip%25253Abrian@freeswitch.org>
> <mailto:sip%253Abrian@freeswitch.org<sip%25253Abrian@freeswitch.org> <mailto:sip%25253Abrian@freeswitch.org<sip%2525253Abrian@freeswitch.org>
> > >> > > >> > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org>> > > <mailto:Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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 > > > > > > > > > > -- > > 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>
> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> > <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com> <mailto:MSN%25253Aanthony_minessale@hotmail.com<MSN%2525253Aanthony_minessale@hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
> <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com> <mailto:PAYPAL%25253Aanthony.minessale@gmail.com<PAYPAL%2525253Aanthony.minessale@gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> > <http://irc.freenode.net> #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> > <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org> <mailto:sip%25253A888@conference.freeswitch.org<sip%2525253A888@conference.freeswitch.org>
> > iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > > <http://iax:guest@conference.freeswitch.org/888> > > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org> <mailto:googletalk%25253Aconf%25252B888@conference.freeswitch.org<googletalk%2525253Aconf%2525252B888@conference.freeswitch.org>
> > pstn:213-799-1400 > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > <mailto: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> > <mailto: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 > > > > > -- > 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>
> <mailto:MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <mailto:MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_minessale@hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <mailto:PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <mailto:sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> > <http://iax:guest@conference.freeswitch.org/888> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@conference.freeswitch.org>
> pstn:213-799-1400 > > >
------------------------------------------------------------------------ > > _______________________________________________ > 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
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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>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com><mailto: PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org><mailto: sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 < http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org><mailto: googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400
------------------------------------------------------------------------
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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
Anthony Minessale napsal(a):
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414 <http://212.20.119.45:31414> codec: 8 ms: 20
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065 <http://23@192.168.1.128:5065>] 192.168.1.128:21322->192.168.1.101\ :49152 codec: 8 ms: 20
Are you setting something in your config to use 0.0.0.0 <http://0.0.0.0> as the media IP? Perhaps putting the real ip in ext-rtp-ip and setting the local one to 0.0.0.0 <http://0.0.0.0> ?
Exactly as you think, Anthony. I set public IP of my network as ext-rtp-ip and 0.0.0.0 as local - I think it tell to FreeSWITCH to bind to all local IPs (currently only one exists, but for th future - I use it in all my servers..) Do I something wrong? Best regards, kokoska.rokoska
putting in 0.0.0.0 as the rtp-ip is unsupported. Try putting the local ip of the box in there instead. Also, we have a special variable that we set with the guessed local ip that you can use: $${local_ip_v4} <-- put this where 0.0.0.0 is This variable is replaced before the startup is complete so the final config will contain the actual ip. On Mon, Apr 21, 2008 at 1:47 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414 <http://212.20.119.45:31414> codec: 8 ms: 20
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065 <http://23@192.168.1.128:5065>] 192.168.1.128:21322->192.168.1.101\ :49152 codec: 8 ms: 20
Are you setting something in your config to use 0.0.0.0 <http://0.0.0.0> as the media IP? Perhaps putting the real ip in ext-rtp-ip and setting the local one to 0.0.0.0 <http://0.0.0.0> ?
Exactly as you think, Anthony. I set public IP of my network as ext-rtp-ip and 0.0.0.0 as local - I think it tell to FreeSWITCH to bind to all local IPs (currently only one exists, but for th future - I use it in all my servers..)
Do I something wrong?
Best regards,
kokoska.rokoska
_______________________________________________ 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/ 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
Anthony Minessale napsal(a):
putting in 0.0.0.0 <http://0.0.0.0> as the rtp-ip is unsupported. Try putting the local ip of the box in there instead.
Thank you very much, Anthony, for notable info! I didn't know that.
Also, we have a special variable that we set with the guessed local ip that you can use: $${local_ip_v4} <-- put this where 0.0.0.0 <http://0.0.0.0> is
This variable is replaced before the startup is complete so the final config will contain the actual ip.
I replaced 0.0.0.0 with local IP address and than with $${local_ip_v4} and both works OK :-) So my trouble si solved, thank you once more! BTW: I'm still not sure if it is desirable behaviour - if FreeSWITCH properly replaces misconfigured IP address but than uses "invalid" (twice the same) port... Best regards, kokoska.rokoska
On Mon, Apr 21, 2008 at 1:47 PM, kokoska rokoska <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
Anthony Minessale napsal(a): > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414 <http://212.20.119.45:31414> > <http://212.20.119.45:31414> codec: 8 ms: 20 > > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065 <http://23@192.168.1.128:5065> > <http://23@192.168.1.128:5065>] 192.168.1.128:21322->192.168.1.101\ > :49152 codec: 8 ms: 20 > > > Are you setting something in your config to use 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> > as the media IP? > Perhaps putting the real ip in ext-rtp-ip and setting the local one to > 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> ? > >
Exactly as you think, Anthony. I set public IP of my network as ext-rtp-ip and 0.0.0.0 <http://0.0.0.0> as local - I think it tell to FreeSWITCH to bind to all local IPs (currently only one exists, but for th future - I use it in all my servers..)
Do I something wrong?
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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------------------------------------------------
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freeswitch uses unique ip/port pairs so freeswitch sees that was 2 different addresses the 0.0.0.0 was considered another ip. 0.0.0.0:1234 1.2.3.4:1234 We will maybe add code to make 0.0.0.0 fatal input as rtp-ip to avoid this issue. On Mon, Apr 21, 2008 at 3:26 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
putting in 0.0.0.0 <http://0.0.0.0> as the rtp-ip is unsupported. Try putting the local ip of the box in there instead.
Thank you very much, Anthony, for notable info! I didn't know that.
Also, we have a special variable that we set with the guessed local ip that you can use: $${local_ip_v4} <-- put this where 0.0.0.0 <http://0.0.0.0> is
This variable is replaced before the startup is complete so the final config will contain the actual ip.
I replaced 0.0.0.0 with local IP address and than with $${local_ip_v4} and both works OK :-) So my trouble si solved, thank you once more!
BTW: I'm still not sure if it is desirable behaviour - if FreeSWITCH properly replaces misconfigured IP address but than uses "invalid" (twice the same) port...
Best regards,
kokoska.rokoska
On Mon, Apr 21, 2008 at 1:47 PM, kokoska rokoska <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
Anthony Minessale napsal(a): > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414 <http://212.20.119.45:31414> > <http://212.20.119.45:31414> codec: 8 ms: 20 > > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065 <http://23@192.168.1.128:5065> > <http://23@192.168.1.128:5065>] 192.168.1.128:21322
->192.168.1.101\
> :49152 codec: 8 ms: 20 > > > Are you setting something in your config to use 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> > as the media IP? > Perhaps putting the real ip in ext-rtp-ip and setting the local one to > 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> ? > >
Exactly as you think, Anthony. I set public IP of my network as ext-rtp-ip and 0.0.0.0 <http://0.0.0.0> as local - I think it tell to FreeSWITCH to bind to all local IPs (currently only
one
exists, but for th future - I use it in all my servers..)
Do I something wrong?
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
-- 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>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400
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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
I forgot to mention the word "auto" should also suffice to auto choose the ip. I added a patch to outlaw 0.0.0.0 going forward. On Mon, Apr 21, 2008 at 3:42 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
freeswitch uses unique ip/port pairs so freeswitch sees that was 2 different addresses the 0.0.0.0 was considered another ip.
0.0.0.0:1234 1.2.3.4:1234
We will maybe add code to make 0.0.0.0 fatal input as rtp-ip to avoid this issue.
On Mon, Apr 21, 2008 at 3:26 PM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Anthony Minessale napsal(a):
putting in 0.0.0.0 <http://0.0.0.0> as the rtp-ip is unsupported. Try putting the local ip of the box in there instead.
Thank you very much, Anthony, for notable info! I didn't know that.
Also, we have a special variable that we set with the guessed local ip that you can use: $${local_ip_v4} <-- put this where 0.0.0.0 <http://0.0.0.0> is
This variable is replaced before the startup is complete so the final config will contain the actual ip.
I replaced 0.0.0.0 with local IP address and than with $${local_ip_v4} and both works OK :-) So my trouble si solved, thank you once more!
BTW: I'm still not sure if it is desirable behaviour - if FreeSWITCH properly replaces misconfigured IP address but than uses "invalid" (twice the same) port...
Best regards,
kokoska.rokoska
On Mon, Apr 21, 2008 at 1:47 PM, kokoska rokoska <kokoska.rokoska@post.cz <mailto:kokoska.rokoska@post.cz>> wrote:
Anthony Minessale napsal(a): > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322->212.20.119.45:31414 <http://212.20.119.45:31414> > <http://212.20.119.45:31414> codec: 8 ms: 20 > > 2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp() > AUDIO RTP [sofia/default/0002*23@192.168.1.128:5065 <http://23@192.168.1.128:5065> > <http://23@192.168.1.128:5065>] 192.168.1.128:21322
->192.168.1.101\
> :49152 codec: 8 ms: 20 > > > Are you setting something in your config to use 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> > as the media IP? > Perhaps putting the real ip in ext-rtp-ip and setting the local one to > 0.0.0.0 <http://0.0.0.0> <http://0.0.0.0> ? > >
Exactly as you think, Anthony. I set public IP of my network as ext-rtp-ip and 0.0.0.0 <http://0.0.0.0> as local - I think it tell to FreeSWITCH to bind to all local IPs (currently
only one
exists, but for th future - I use it in all my servers..)
Do I something wrong?
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
-- 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>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> <mailto:PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> <mailto:sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
pstn:213-799-1400
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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
-- 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
Anthony Minessale napsal(a):
freeswitch uses unique ip/port pairs so freeswitch sees that was 2 different addresses the 0.0.0.0 <http://0.0.0.0> was considered another ip.
0.0.0.0:1234 <http://0.0.0.0:1234> 1.2.3.4:1234 <http://1.2.3.4:1234>
Yes, I catch it from your previous answer :-)
We will maybe add code to make 0.0.0.0 <http://0.0.0.0> fatal input as rtp-ip to avoid this issue.
IMO it will be very useful and can avoid some unnecessary questions, like mine were :-) Thanks again, Anthony, for very notable help! Best regards, kokoska.rokoska
participants (5)
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Anthony Minessale
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Brian West
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kokoska rokoska
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Mark Campbell-Smith
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Wasim Baig