Thanks for all suggestions. Ufortunately I cannot get it working.
Seems like packets are not coming to phone behind nat (freeswitch is on public
ip).
When registering I can see multiple notify retries like this:
send 802 bytes to udp/[10.99.10.6]:5060 at 10:49:31.762605:
------------------------------------------------------------------------
NOTIFY sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 212.235.180.41:5080;rport;branch=z9hG4bKtNStS2gtr8DNr
Max-Forwards: 70
From: <sip:[email protected]>;tag=veSr4DmgmFHjr
To: <sip:[email protected]>
Call-ID: cec2b00b-5814-122c-f981-000fea488302
CSeq: 109587536 NOTIFY
Contact: <sip:[email protected]:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10924M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, refer
Subscription-State: terminated;timeout
Content-Type: application/simple-message-summary
Content-Length: 93
Messages-Waiting: yes
Message-Account: sip:[email protected]
Voice-Message: 3/0 (0/0)
I've opened necessary ports and I've defined custom rtp port range (which goes
trough).
Does nat should really just work if you register on external profile via port
5080? This is
what I'm doing now.
The phone on lan is a nokia N95 configured like described here (using port
5080):
http://wiki.freeswitch.org/wiki/Nokia_N95
Phone shows registered message, but it takes like a half minute to register,
when I'm on home network this happens in a second.
On Wed, 07 Jan 2009 12:39:45 +0100, Peter P GMX <[email protected]> wrote:
> Generally speaking you will need to open an UPD port range for the RTP
> stream. This can be configured on FS. Eg. we use 12000-13000 on our
> system.
> Then If you do not hear any sound you may put
>
> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>
>
> in your external and internal profile, if FS is natted.
>
> Best regards
> Peter
>
>
> kriko schrieb:
>> Hello!
>>
>> Yesterday I've successfully placed a call between two different domains:
>> originate sofia/default/[email protected]
>> &bridge(sofia/gateway/212.235.180.41/1001)
>>
>> I didn't hear any audio, but it was kinda working. Today I investigated
>> this more deep and found some issues.
>> FS with 212.235.180.41 is a public computer with firewall, but open TCP
>> and UDP 5060, 5080 ports. Freeswitch on this machine
>> uses default configuration.
>>
>> FS with 10.99.8.221 is a lan computer in a different place, this is
>> where I would like to start a call, the other way would
>> be probably too much difficult for now. I've added a gateway entry to
>> this one:
>> http://pastebin.com/m2174ead
>>
>> Calling from 10.99.8.221 (for e.g. using softphone at ext. 1003) to
>> 212.235.180.41 (ext. 1001 for e.g.) works. Both end
>> answers, however I cannot hear audio coming trough. When testing I'm at
>> the computer which is behind a lan, so I'm
>> capturing music as audio source on the other side.
>>
>> Are there any other ports I should open on public computer?
>> With wireshark on the computer behind a lan, I can see RTP going away
>> to 212.235.180.41, but not the other way.
>>
>> There are also issues when e.g. terminating a call on public computer,
>> fs on the other end will never terminate the call since
>> SIP messages cannot reach the computer behind lan I guess, but this is
>> second problem.
>>
>>
>>
>
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--
kriko
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