Re: [Freeswitch-users] how to config FS with two net interface?
Ok then something is broken badly... and that makes NO sense. Because it works on my box. /b On Oct 27, 2009, at 8:34 PM, Lei Tang wrote: Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I set the ext-rtp-ip, it works fine. 2009/10/28 Brian West you should have just needed to set the rtp-ip and the sip-ip on both profiles to their values and it would have worked fine... their would have been no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: > Thanks Eliot, It works. ___ 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 -- Lei.Tang lei.tl...@gmail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to config FS with two net interface?
Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I set the ext-rtp-ip, it works fine. 2009/10/28 Brian West > you should have just needed to set the rtp-ip and the sip-ip on both > profiles to their values and it would have worked fine... their would > have been no need to set the ext-*-ip equiv. > > /b > > On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: > > > Thanks Eliot, It works. > > > ___ > 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 > -- Lei.Tang lei.tl...@gmail.com ___ 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
Re: [Freeswitch-users] how to config FS with two net interface?
you should have just needed to set the rtp-ip and the sip-ip on both profiles to their values and it would have worked fine... their would have been no need to set the ext-*-ip equiv. /b On Oct 27, 2009, at 8:14 PM, Lei Tang wrote: > Thanks Eliot, It works. ___ 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
Re: [Freeswitch-users] how to config FS with two net interface?
Thanks Eliot, It works.
2009/10/27 Eliot Gable
>
> Try setting ext-rtp-ip and ext-sip-ip on both profiles.
>
> On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang wrote:
> > Hi all, I run FS on a machine with two net interface, each interface has
> a
> > ip addr, one of the them connect to public network(has ip addr A), the
> > other connect to a private network(has ip addr B), FS server as a SIP
> > server for public through A, all outbound call will bridge to a
> softswitch
> > in private network through B. here is my sofia config file and diaplan
> > config:
> >
> > sofia internal.xml
> >
> >
> >
> >
> >
> > sofia external.xml
> >
> >
> >
> >
> >
> > dialplan
> > ..
> >
> >
> >
> > >
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> > > data="effective_caller_id_number=xxx"/>
> > > data="sofia/external/${destination_numb...@x"/>
> >
> >
> > .
> >
> > then call seq is
> > sipAgent --> [internal -->(bridge)-->external] -->softswith
> > FREESWITCH
> >
> > the question is, when sipAgent make a outbound call, FS can't recevie the
> > caller's up audio stream, I traced the SIP packets, found that FS has
> return
> > addr B in SDP when ack the invite request from sipAgent, the ack packet
> is
> > ===
> > SIP/2.0 183 Session Progress
> > Via: SIP/2.0/UDP
> >
> x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> > From: "1000" ;tag=cb4d3c4e
> > To: "65960581" ;tag=DtvSc0QX01yKN
> > Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> > CSeq: 2 INVITE
> > Contact:
> > User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> > Accept: application/sdp
> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
> > REFER, UPDATE, REGISTER, INFO, PUBLISH
> > Supported: timer, precondition, path, replaces
> > Allow-Events: talk, presence, dialog, call-info, sla,
> > include-session-description, presence.winfo, message-summary, refer
> > Content-Type: application/sdp
> > Content-Disposition: session
> > Content-Length: 245
> >
> > v=0
> > o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip
> addr
> > of the adapter connect to the private network
> > s=FreeSWITCH
> > c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
> > private network
> > t=0 0
> > m=audio 31066 RTP/AVP 0 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > I think FS should return A in SDP, not the external binding addr (B),
> does
> > somebody known how to solve this problem?
> >
> > --
> > Lei.Tang
> > lei.tl...@gmail.com
> >
> > ___
> > 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
> >
> >
>
>
>
> --
> Eliot Gable
>
> "We do not inherit the Earth from our ancestors: we borrow it from our
> children." ~David Brower
>
> "I decided the words were too conservative for me. We're not borrowing
> from our children, we're stealing from them--and it's not even
> considered to be a crime." ~David Brower
>
> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
> live; not live to eat.) ~Marcus Tullius Cicero
>
> ___
> 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
>
--
Lei.Tang
lei.tl...@gmail.com
___
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: [Freeswitch-users] how to config FS with two net interface?
Try setting ext-rtp-ip and ext-sip-ip on both profiles.
On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang wrote:
> Hi all, I run FS on a machine with two net interface, each interface has a
> ip addr, one of the them connect to public network(has ip addr A), the
> other connect to a private network(has ip addr B), FS server as a SIP
> server for public through A, all outbound call will bridge to a softswitch
> in private network through B. here is my sofia config file and diaplan
> config:
>
> sofia internal.xml
>
>
>
>
>
> sofia external.xml
>
>
>
>
>
> dialplan
> ..
>
>
>
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
> data="effective_caller_id_number=xxx"/>
> data="sofia/external/${destination_numb...@x"/>
>
>
> .
>
> then call seq is
> sipAgent --> [internal -->(bridge)-->external] -->softswith
> FREESWITCH
>
> the question is, when sipAgent make a outbound call, FS can't recevie the
> caller's up audio stream, I traced the SIP packets, found that FS has return
> addr B in SDP when ack the invite request from sipAgent, the ack packet is
> ===
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
> From: "1000" ;tag=cb4d3c4e
> To: "65960581" ;tag=DtvSc0QX01yKN
> Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
> CSeq: 2 INVITE
> Contact:
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
> REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 245
>
> v=0
> o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip addr
> of the adapter connect to the private network
> s=FreeSWITCH
> c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the
> private network
> t=0 0
> m=audio 31066 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> I think FS should return A in SDP, not the external binding addr (B), does
> somebody known how to solve this problem?
>
> --
> Lei.Tang
> lei.tl...@gmail.com
>
> ___
> 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
>
>
--
Eliot Gable
"We do not inherit the Earth from our ancestors: we borrow it from our
children." ~David Brower
"I decided the words were too conservative for me. We're not borrowing
from our children, we're stealing from them--and it's not even
considered to be a crime." ~David Brower
"Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to
live; not live to eat.) ~Marcus Tullius Cicero
___
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
[Freeswitch-users] how to config FS with two net interface?
Hi all, I run FS on a machine with two net interface, each interface has a ip addr, one of the them connect to public network(has ip addr A), the other connect to a private network(has ip addr B), FS server as a SIP server for public through A, all outbound call will bridge to a softswitch in private network through B. here is my sofia config file and diaplan config: sofia internal.xml sofia external.xml dialplan .. . then call seq is sipAgent --> [internal -->(bridge)-->external] -->softswith FREESWITCH the question is, when sipAgent make a outbound call, FS can't recevie the caller's up audio stream, I traced the SIP packets, found that FS has return addr B in SDP when ack the invite request from sipAgent, the ack packet is === SIP/2.0 183 Session Progress Via: SIP/2.0/UDP x:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208 From: "1000" ;tag=cb4d3c4e To: "65960581" ;tag=DtvSc0QX01yKN Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI. CSeq: 2 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;wrong this is the ip addr of the adapter connect to the private network s=FreeSWITCH c=IN IP4 B ;wrong this is the ip addr of the adapter connect to the private network t=0 0 m=audio 31066 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 I think FS should return A in SDP, not the external binding addr (B), does somebody known how to solve this problem? -- Lei.Tang lei.tl...@gmail.com ___ 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
