can't dial from IPv6 to IPv6
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 9999 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.
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 9999 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.
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 <brian@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 9999 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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And you're sure you have that IP on your interface? William or Jason care to let me know if latest SVN IPv6 works for you? /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
On Mon, Oct 19, 2009 at 5:13 PM, Brian West <brian@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
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
IP should be OK: [root@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: 1007@franta.openstage.net Contact: 1007 <sip:1007@[2000:2::1002]:5062;transport=udp> but when it comes to connecting the call: EXECUTE sofia/internal-ipv6/1006@franta.openstage.net bridge(user/1007@franta.openstage.net) 2009-10-20 07:36:12.749270 [DEBUG] switch_ivr_originate.c:1027 variable string 0 = [presence_id=1007@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 <brian@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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On 20/10/09 07:53 +0200, ineya ineya wrote:
IP should be OK:
[root@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
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
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 <dwhite@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
Well how are you trying to dial users? /b On Oct 20, 2009, at 9:16 AM, ineya ineya wrote:
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.
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 <brian@freeswitch.org> wrote:
Well how are you trying to dial users?
/b
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/user@domain /b On Oct 20, 2009, at 10:21 AM, ineya ineya wrote:
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.
Hmm, I didn't about this. so in directory/default.xml I have: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> and the modified version for IPv6 would be ?... <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/> On Tue, Oct 20, 2009 at 5:34 PM, Brian West <brian@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/user@domain
/b
ineya ineya <ineyaa@gmail.com> wrote:
Hmm, I didn't about this.
so in directory/default.xml I have: <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
and the modified version for IPv6 would be ?... <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_user}@${dialed_domain})}"/>
Yes. Are you suggesting it didn't work?
Sort of. I have different error now: 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal-ipv6/1006@franta.openstage.net! 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup sofia/internal-ipv6/sip:1007@[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?
ineya ineya <ineyaa@gmail.com> wrote:
Sort of. I have different error now:
2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal-ipv6/1006@franta.openstage.net! 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup sofia/internal-ipv6/sip:1007@[2000:2::1002]:5062 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
I suspect the codec negotiation. Make sure that both ends are offering a common codec.
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 <jason@jasonjgw.net> wrote:
I suspect the codec negotiation. Make sure that both ends are offering a common codec.
What are you using to make this call? /b On Oct 21, 2009, at 6:58 AM, ineya ineya wrote:
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?
This appears to be some sort of ice implementation? We don't support sip ice at this time. Mike On Oct 21, 2009, at 7:58 AM, ineya ineya wrote:
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 <jason@jasonjgw.net> wrote:
I suspect the codec negotiation. Make sure that both ends are offering a common codec.
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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 <mike@jerris.com> wrote:
This appears to be some sort of ice implementation? We don't support sip ice at this time.
participants (5)
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Brian West
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Dan White
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ineya ineya
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Jason White
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Michael Jerris