Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Ivan C Myrvold
Yeah, the call arrives nicely now, but the audio is only 1-way.
I have port forwarded all the ports in my router to FreeSwitch, but  
still experiences just oneway audio, from FreeSwitch to the DID caller.
Looks like the RTP is not forwarded correctly to FreeSwitch.
Could it be my Linksys router, which is not forwarding RTP ports  
correctly, or is there still a piece in FreeSwitch I have missed?

Ivan

Den 20. juni. 2008 kl. 22:52 skrev Brian West:

 Good to know it snapped into place now! :P

 /b
 On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:

 This is all making sense to me now.


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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Ivan C Myrvold
I put this into internal.xml, and this seems to do the trick:

 param name=ext-rtp-ip value=$${external_rtp_ip}/
 param name=ext-sip-ip value=$${external_sip_ip}/

Ivan

Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold:

 Yeah, the call arrives nicely now, but the audio is only 1-way.
 I have port forwarded all the ports in my router to FreeSwitch, but
 still experiences just oneway audio, from FreeSwitch to the DID  
 caller.
 Looks like the RTP is not forwarded correctly to FreeSwitch.
 Could it be my Linksys router, which is not forwarding RTP ports
 correctly, or is there still a piece in FreeSwitch I have missed?

 Ivan

 Den 20. juni. 2008 kl. 22:52 skrev Brian West:

 Good to know it snapped into place now! :P

 /b
 On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:

 This is all making sense to me now.


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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Brian West
Let me guess.. you have phones behind the same NAT talking to this  
profile also?

If that is the case you'll need to create a new profile with ext-sip- 
ip and ext-rtp-ip set for the outside world to talk to.. then take  
those settings off the profile your phones behind the nat talk to  
freeswitch.

/b


On Jun 21, 2008, at 10:56 AM, Ivan C Myrvold wrote:

 I put this into internal.xml, and this seems to do the trick:

 param name=ext-rtp-ip value=$${external_rtp_ip}/
 param name=ext-sip-ip value=$${external_sip_ip}/

 Ivan

 Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold:

 Yeah, the call arrives nicely now, but the audio is only 1-way.
 I have port forwarded all the ports in my router to FreeSwitch, but
 still experiences just oneway audio, from FreeSwitch to the DID
 caller.
 Looks like the RTP is not forwarded correctly to FreeSwitch.
 Could it be my Linksys router, which is not forwarding RTP ports
 correctly, or is there still a piece in FreeSwitch I have missed?

 Ivan

 Den 20. juni. 2008 kl. 22:52 skrev Brian West:

 Good to know it snapped into place now! :P

 /b
 On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:

 This is all making sense to me now.


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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-21 Thread Brian West
http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios

Please review Scenario 1 the bold text.  ;)

/b

On Jun 21, 2008, at 10:56 AM, Ivan C Myrvold wrote:

 I put this into internal.xml, and this seems to do the trick:

 param name=ext-rtp-ip value=$${external_rtp_ip}/
 param name=ext-sip-ip value=$${external_sip_ip}/

 Ivan

 Den 21. juni. 2008 kl. 15:47 skrev Ivan C Myrvold:

 Yeah, the call arrives nicely now, but the audio is only 1-way.
 I have port forwarded all the ports in my router to FreeSwitch, but
 still experiences just oneway audio, from FreeSwitch to the DID
 caller.
 Looks like the RTP is not forwarded correctly to FreeSwitch.
 Could it be my Linksys router, which is not forwarding RTP ports
 correctly, or is there still a piece in FreeSwitch I have missed?

 Ivan

 Den 20. juni. 2008 kl. 22:52 skrev Brian West:

 Good to know it snapped into place now! :P

 /b
 On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:

 This is all making sense to me now.


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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
As I understand the ACL, it only controls which external machines with  
the IP address range given in the acl.conf.xml are allowed into  
FreeSwitch.
So this doesn't help me much with the DID, as FreeSwitch sends a  
Proxy Authentication Required SIP message back to the Voxbone server.
How can I configure FreeSwitch to allow the call into FreeSwitch  
without proxy authentication?

Ivan

Den 18. juni. 2008 kl. 09:25 skrev Ivan C Myrvold:

 I added to my acl.conf.xml this
 http://pastebin.freeswitch.org/4640

 and this to external.xml
   param name=apply-inbound-acl value=voxbone/

 When I start freeswitch, I see that the settings are used:

 2008-06-18 09:11:09 [CONSOLE] switch_core.c:769
 switch_load_network_lists() Created ip list voxbone default (deny)
 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791
 switch_load_network_lists() Adding 81.201.82.0/24 (allow) to list
 voxbone
 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791
 switch_load_network_lists() Adding 81.201.86.0/24 (allow) to list
 voxbone
 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791
 switch_load_network_lists() Adding 81.201.83.0/24 (allow) to list
 voxbone
 2008-06-18 09:11:09 [CONSOLE] switch_core.c:791
 switch_load_network_lists() Adding 81.201.84.0/24 (allow) to list
 voxbone

 But still I get the same result as before:

 http://pastebin.freeswitch.org/4639

 The list of IP addresses are at
 http://www.voxbone.com/members/faq-technical.jsf , and I am not
 absolutely sure I put this into the acl.conf.xml list correctly?

 Ivan

 Den 17. juni. 2008 kl. 16:08 skrev Brian West:

 You'll need to setup an ACL to let them in without authentication.
 Look at the Asterisk to FreeSWITCH section on the wiki.

 http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

 /b

 On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote:

 I have used Freeswitch with a DID from Voxbone working on IAX, and
 that have been working so well. But now Voxbone will discontinue the
 IAX service, so I have to get DID working on SIP, and I am wondering
 if that will work at all in my configuration.

 Freeswitch is behind nat, so when I get a call from Voxbone, see 
 http://pastebin.freeswitch.org/4629
 ,
 it will come in on port 5060. Have you covered this situation in the
 documentation?

 Ivan

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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Ivan,
Do you have your outbound registration to voxbone on the default/ 
internal profile?  If so then your acl's might be wrong.  The one sure  
fire way to do this is to just setup another profile without auth on a  
different port and run with that.

/b

On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote:

 As I understand the ACL, it only controls which external machines with
 the IP address range given in the acl.conf.xml are allowed into
 FreeSwitch.
 So this doesn't help me much with the DID, as FreeSwitch sends a
 Proxy Authentication Required SIP message back to the Voxbone  
 server.
 How can I configure FreeSwitch to allow the call into FreeSwitch
 without proxy authentication?

 Ivan

Brian West
sip:[EMAIL PROTECTED]




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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Let me guess they only send to port 5060?  If you have your ACL's  
setup correctly those two IP's will be let in without auth.

If you have the profile that runs on 5060 on your FreeSWITCH box with  
auth-calls=true

Add this param name=apply-inbound-acl value=voxbone/ to that  
profile

list name=voxbone default=deny
node type=allow cidr=81.201.82.20/32/
node type=allow cidr=81.201.82.21/32/
/list

This should allow them thru without auth.  If you have problems with  
that add 81.201.82.0/24 to the list and see if that makes a difference.

/b



On Jun 20, 2008, at 11:28 AM, Ivan C Myrvold wrote:

 I do not have outbound registation to Voxbone, because Voxbone is  
 only incoming. I am not registrating Voxbone at all.
 In their FAQ, they have how to configure for Asterisk:

 [81.201.82.20]
 host = 81.201.82.20
 type = friend
 insecure = very
 context = your-context
 canreinvite=no

 [81.201.82.21]
 host = 81.201.82.21
 type = friend
 insecure = very
 context = your-context
 canreinvite=no
 ...
 ( Do it for all voxbone IPs address listed above )

 Den 20. juni. 2008 kl. 17:33 skrev Brian West:

 Ivan,
  Do you have your outbound registration to voxbone on the default/
 internal profile?  If so then your acl's might be wrong.  The one  
 sure
 fire way to do this is to just setup another profile without auth  
 on a
 different port and run with that.

 /b

 On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote:

 As I understand the ACL, it only controls which external machines  
 with
 the IP address range given in the acl.conf.xml are allowed into
 FreeSwitch.
 So this doesn't help me much with the DID, as FreeSwitch sends a
 Proxy Authentication Required SIP message back to the Voxbone
 server.
 How can I configure FreeSwitch to allow the call into FreeSwitch
 without proxy authentication?

 Ivan

 Brian West
 sip:[EMAIL PROTECTED]




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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Ivan C Myrvold
Ah, I put the apply-inbound-acl in the wrong XML file. I put it in  
the external.xml.
When I instead put it in the internal.xml, I got it working. That is  
of course the correct place, because it is bound to port 5060, and the  
inbound from Voxbone comes in on port 5060, as you correctly guessed.

This is all making sense to me now.

Ivan

Den 20. juni. 2008 kl. 19:17 skrev Brian West:

 Let me guess they only send to port 5060?  If you have your ACL's
 setup correctly those two IP's will be let in without auth.

 If you have the profile that runs on 5060 on your FreeSWITCH box with
 auth-calls=true

 Add this param name=apply-inbound-acl value=voxbone/ to that
 profile

 list name=voxbone default=deny
 node type=allow cidr=81.201.82.20/32/
 node type=allow cidr=81.201.82.21/32/
 /list

 This should allow them thru without auth.  If you have problems with
 that add 81.201.82.0/24 to the list and see if that makes a  
 difference.

 /b



 On Jun 20, 2008, at 11:28 AM, Ivan C Myrvold wrote:

 I do not have outbound registation to Voxbone, because Voxbone is
 only incoming. I am not registrating Voxbone at all.
 In their FAQ, they have how to configure for Asterisk:

 [81.201.82.20]
 host = 81.201.82.20
 type = friend
 insecure = very
 context = your-context
 canreinvite=no

 [81.201.82.21]
 host = 81.201.82.21
 type = friend
 insecure = very
 context = your-context
 canreinvite=no
 ...
 ( Do it for all voxbone IPs address listed above )

 Den 20. juni. 2008 kl. 17:33 skrev Brian West:

 Ivan,
 Do you have your outbound registration to voxbone on the default/
 internal profile?  If so then your acl's might be wrong.  The one
 sure
 fire way to do this is to just setup another profile without auth
 on a
 different port and run with that.

 /b

 On Jun 20, 2008, at 3:20 AM, Ivan C Myrvold wrote:

 As I understand the ACL, it only controls which external machines
 with
 the IP address range given in the acl.conf.xml are allowed into
 FreeSwitch.
 So this doesn't help me much with the DID, as FreeSwitch sends a
 Proxy Authentication Required SIP message back to the Voxbone
 server.
 How can I configure FreeSwitch to allow the call into FreeSwitch
 without proxy authentication?

 Ivan

 Brian West
 sip:[EMAIL PROTECTED]




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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-20 Thread Brian West
Good to know it snapped into place now! :P

/b
On Jun 20, 2008, at 3:49 PM, Ivan C Myrvold wrote:

 This is all making sense to me now.


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Re: [Freeswitch-users] Can I get SIP DID working?

2008-06-18 Thread Ivan C Myrvold
I added to my acl.conf.xml this
http://pastebin.freeswitch.org/4640

and this to external.xml
 param name=apply-inbound-acl value=voxbone/

When I start freeswitch, I see that the settings are used:

2008-06-18 09:11:09 [CONSOLE] switch_core.c:769  
switch_load_network_lists() Created ip list voxbone default (deny)
2008-06-18 09:11:09 [CONSOLE] switch_core.c:791  
switch_load_network_lists() Adding 81.201.82.0/24 (allow) to list  
voxbone
2008-06-18 09:11:09 [CONSOLE] switch_core.c:791  
switch_load_network_lists() Adding 81.201.86.0/24 (allow) to list  
voxbone
2008-06-18 09:11:09 [CONSOLE] switch_core.c:791  
switch_load_network_lists() Adding 81.201.83.0/24 (allow) to list  
voxbone
2008-06-18 09:11:09 [CONSOLE] switch_core.c:791  
switch_load_network_lists() Adding 81.201.84.0/24 (allow) to list  
voxbone

But still I get the same result as before:

http://pastebin.freeswitch.org/4639

The list of IP addresses are at
http://www.voxbone.com/members/faq-technical.jsf , and I am not  
absolutely sure I put this into the acl.conf.xml list correctly?

Ivan

Den 17. juni. 2008 kl. 16:08 skrev Brian West:

 You'll need to setup an ACL to let them in without authentication.
 Look at the Asterisk to FreeSWITCH section on the wiki.

 http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

 /b

 On Jun 17, 2008, at 12:31 AM, Ivan C Myrvold wrote:

 I have used Freeswitch with a DID from Voxbone working on IAX, and
 that have been working so well. But now Voxbone will discontinue the
 IAX service, so I have to get DID working on SIP, and I am wondering
 if that will work at all in my configuration.

 Freeswitch is behind nat, so when I get a call from Voxbone, see 
 http://pastebin.freeswitch.org/4629
 ,
 it will come in on port 5060. Have you covered this situation in the
 documentation?

 Ivan

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 sip:[EMAIL PROTECTED]




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