Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-21 Thread ineya ineya
Sort of. I have different error now:

2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer
sofia/internal-ipv6/1...@franta.openstage.net!
2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup
sofia/internal-ipv6/sip:1...@[2000:2::1002]:5062 [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]

Here is the complete log:
http://pastebin.freeswitch.org/10782

 Yes. Are you suggesting it didn't work?



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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-21 Thread ineya ineya
Codecs are fine. I spent much time experimenting with codecs and
completely missed, that freeswitch is modifiyng the SDP record.

When phone A is making a call the SDP contains candidate media attributes:

a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host
a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host

But when freeswitch makes the INVITE on phone B, these 2 are missing
and phone is looking for it, so the INVITE gets rejected by phone with
448 Not acceptable here

So the question is, how can I make the freeswitch to pass these
candidate media attributes?

On Wed, Oct 21, 2009 at 8:58 AM, Jason White ja...@jasonjgw.net wrote:


 I suspect the codec negotiation. Make sure that both ends are offering a
 common codec.


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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-21 Thread ineya ineya
the phone is called OpenStage

On Wed, Oct 21, 2009 at 4:25 PM, Brian West br...@freeswitch.org wrote:
 What are you using to make this call?

 /b


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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-21 Thread ineya ineya
Yes, I think you are right, the sip stack comes from company called
M5T as far as I know, and there are references to ICE on their site.

On Wed, Oct 21, 2009 at 7:29 PM, Michael Jerris m...@jerris.com wrote:
 This appears to be some sort of ice implementation?  We don't support
 sip ice at this time.


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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-20 Thread ineya ineya
No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.

Is there a softphone, which you know works with freeswitch on IPv6?
Maybe the error is in phones I'm currently using.

On Tue, Oct 20, 2009 at 3:34 PM, Dan White dwh...@olp.net wrote:

 This address caught my eye. Although valid, a /32 address assignment is
 atypical. Is that what you intended? Does assigning a /64 address on both
 ends make any difference?

 --
 Dan White


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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-20 Thread ineya ineya
By numbers, I have 2 numbers registered, so I dial 1006 or 1007.
This worked for IPv4, so I haven't thought about doing it differently for IPv6.
SIP messages looked OK to me - you just gave me an idea to compare SIP
messages for IPv4 and IPv6. I'll try that tommorow.


On Tue, Oct 20, 2009 at 4:21 PM, Brian West br...@freeswitch.org wrote:
 Well how are you trying to dial users?

 /b


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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-20 Thread ineya ineya
Hmm, I didn't about this.

so in directory/default.xml I have:
 param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/

and the modified version for IPv6 would be ?...
 param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_us...@${dialed_domain})}/



On Tue, Oct 20, 2009 at 5:34 PM, Brian West br...@freeswitch.org wrote:
 The defaults will NOT work with ipv6 out of the box because the
 sofia_contact on the directory only looks at the internal profile NOT
 the internal-ipv6 profile... open up the directory default and change
 the sofia_contact to prepend the internal-ipv6/u...@domain

 /b


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[Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-19 Thread ineya ineya
Hi,

I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
port. Directory and dialplan should be default ones.
My phones are able to register when using just IPv4, and I can call from one
to another.

When I try the same thing with IPv6, the call to other phone fails with [
USER_NOT_REGISTERED], however I'm able to call  and hear MOH.
Here you can find the log: http://pastebin.freeswitch.org/10757
Freeswitch is running on 10.80.62.40/2000:2::1
phones are 2000:2::1001 and 2000:2::1002

Am I doing something wrong? Maybe I forgot to configure something, but I
can't see what it is.

Any hints are appreciated.
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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-19 Thread ineya ineya
Here is internal.xml
http://pastebin.freeswitch.org/10766

and internal-ipv6.xml
http://pastebin.freeswitch.org/10767



On Mon, Oct 19, 2009 at 5:13 PM, Brian West br...@freeswitch.org wrote:

 I think you just have it misconfigured because if ipv6 was broken I would
 have William and Jason jumping on me.
 Can you post your xml configs for the profile please?

 /b

 On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:

 Hi,

 I installed the phonenix release, and changed IPv4, IPv6 addresses and SIP
 port. Directory and dialplan should be default ones.
 My phones are able to register when using just IPv4, and I can call from
 one to another.

 When I try the same thing with IPv6, the call to other phone fails with [
 USER_NOT_REGISTERED], however I'm able to call  and hear MOH.
 Here you can find the log: http://pastebin.freeswitch.org/10757
 Freeswitch is running on 10.80.62.40/2000:2::1
 phones are 2000:2::1001 and 2000:2::1002

 Am I doing something wrong? Maybe I forgot to configure something, but I
 can't see what it is.

 Any hints are appreciated.



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Re: [Freeswitch-users] can't dial from IPv6 to IPv6

2009-10-19 Thread ineya ineya
IP should be OK:

[r...@franta /opt/freeswitch]# ifconfig eth1
eth1  Link encap:Ethernet  HWaddr 00:4f:4e:62:ad:83
  inet addr:10.80.62.40  Bcast:10.80.62.255  Mask:255.255.255.0
  inet6 addr: 2000:2::1/32 Scope:Global
  inet6 addr: fe80::24f:4eff:fe62:ad83/64 Scope:Link
  UP BROADCAST RUNNING MULTICAST  MTU:1500  Metric:1
  RX packets:4310768 errors:0 dropped:0 overruns:0 frame:0
  TX packets:2537585 errors:0 dropped:0 overruns:0 carrier:0
  collisions:114862 txqueuelen:1000
  RX bytes:1067582578 (1018.1 MiB)  TX bytes:616134369 (587.5 MiB)
  Interrupt:19 Base address:0x3400

Console debug: http://pastebin.freeswitch.org/10768

When I do sofia status profile internal-ipv6, I can see both phones
as registered:
Call-ID:        9cb5faf2a052867b
User:           1...@franta.openstage.net
Contact:        1007 sip:1...@[2000:2::1002]:5062;transport=udp

but when it comes to connecting the call:

EXECUTE sofia/internal-ipv6/1...@franta.openstage.net
bridge(user/1...@franta.openstage.net)
2009-10-20 07:36:12.749270 [DEBUG] switch_ivr_originate.c:1027
variable string 0 = [presence_id=1...@franta.openstage.net]
2009-10-20 07:36:12.749270 [ERR] switch_ivr_originate.c:1510 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]

@Bill: Can you share your IPv6 test config?

On Tue, Oct 20, 2009 at 3:05 AM, Brian West br...@freeswitch.org wrote:

 Bill has confirmed it works fine in latest trunk.  So i'm not sure what 
 exactly you're doing wrong can you provide me some debug output of what 
 you're doing in your dialplan and the console debug?
 /b
 On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:

 Here is internal.xml
 http://pastebin.freeswitch.org/10766

 and internal-ipv6.xml
 http://pastebin.freeswitch.org/10767


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Re: [Freeswitch-users] dialplan_hunt taking too long

2008-10-10 Thread Ineya Ineya
Thanks for the tip with f8. I did a call from soft phone (x-lite) 1003 to 
extension 5000. 
This is the log:
http://pastebin.freeswitch.org/5791

So it seems, that dialplan_hunt is ok. As you can see, there is a gap ~8 
seconds between line 127 and 128.

The PC is pentium 3, 1.8 GHz, which is pretty idle. It is running Debian etch 
x86 on kernel 2.6.18-6-686. I also made the same test on windows xp, and on 
other notebook running Debian Etch 64bit, and the gap was smaller ( 1-2 
seconds).

I also did run freeswitch using gdb, and after I dialled extension 5000, I 
waited for a few seconds, and then I hit ctrl+c and made backtraces from all 
threads:
http://pastebin.freeswitch.org/5792

sample Wireshark trace on client side when dialling out extension 5000 (client 
is 163.242.69.92, server is 172.16.3.40):
http://pastebin.freeswitch.org/5793

My guess would be the \'while\' inside switch_rtp_set_local_address(), but I 
prefer to hear the expert opinion :-).

Thank you.

- Original message-
Od: \Anthony Minessale\  
Komu:  
Poslaná: 10.10.2008 04:05 
Predmet: Re: [Freeswitch-users] dialplan_hunt taking too long

 
 what extension is it hitting?
 The entire log with debug on would be more useful (press f8 on the console
 and try the call again)
 
 



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