RE: [Asterisk-Users] SPA and NAT traversal
I appreciate everyone's help with setting up an external extension.
Here's a diagram
{SPA2000} - NAT1 - Internet - NAT2 - Asterisk - SIPPhone
SIPPhone is on the same internal subnet as *
NAT2 has a public/staic IP and ports are forwarded to Asterisk
I can successfully do the following:
1. call from SPA2000 - Asterisk VM
2. call from SPA2000 to SIPPhone
3. call from SPA2000 to outside PST as long as I'm using IAX to the ITSP
What I can't do is call from SPA2000 to an outside PSTN if the ITSP is SIP.
When I call the outside phone number I don't hear any ring back or when the
called party answers. If I have RTP DEBUG on at the CLI I don't see any
RTP packets at all so it appears * is outside the media stream.
SIP-PSTN works great so long as the ITSP is being reached by IAX.
This situation exists regardless of the calue for canreinvite.
SIP PEER:
* Name : 202
Secret : Set
MD5Secret: Not set
Context : sip
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox :
LastMsgsSent : -1
Inc. limit : 0
Outg. limit : 0
Dynamic : Yes
Callerid :
Expire : 2749
Expiry : 900
Insecure : no
Nat : Always
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr-IP : 24.6.249.xxx Port 5060
Defaddr-IP : 0.0.0.0 Port 5060
Def. Username: 202
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : UNKNOWN
Useragent: Sipura/SPA2000-2.0.10(c)
Reg. Contact : sip:[EMAIL PROTECTED]:5060
[202] ;test ata
type=friend
username=202
secret=
host=dynamic
nat=no
reinvite=no ;stay in the call
canreinvite=no ;keeps asterisk in the media stream
disallow=all
allow=ulaw
context=sip
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Saturday, April 09, 2005 1:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA and NAT traversal
In your second option using a STUN server would I need to setup my
own STUN server?
No, use FWD or xten's STUN servers.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
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Re: [Asterisk-Users] SPA and NAT traversal
Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... The SPA is behind a NAT and traversing the public IP network to get to the * server. It is successfully registering, thus I can ring a phone registered locally to the * server. I made sure localnet=192.168.2.9/255.255.255.0 (my local cfg for *) and externip=65.87.x.x (which is the public IP of my * server). The * server is behind a NAT as well with the 5060 and 16384-32767 UDP ports open. Based on RTP debug it appears the RTP packets are making it to the * server, the problem is the return address is the internal NAT address of the SPA 192.168.1.100 and not it's public address. Are you willing to share your Martha collection or are you going to keep it to yourself? :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Saturday, April 09, 2005 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA and NAT traversal Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA and NAT traversal
Jim Sturtevant wrote: Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... nat=yes makes Asterisk use the public IP that is inserted by the far side NAT router instead of the private IP the SIP device puts in the packet. Perhaps there is a problem in your sip.conf that is causing the SPA's packets not to match anything. sip show peers will tell you if Asterisk is seeing the public or the private IP of the far end SPA. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... Do I understand your question correctly: You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both devices register, but calls between the devices result in no audio? If that is the case, you can do one of two things: - set canreinvite=no for the devices' sip.conf entries, or - teach both devices to *stop* using their internal IPs for all communications and remove nat=yes from the entry for the SIP device inside NAT2. To set the SPA to give the correct IP, enable STUN, add a STUN server, and say Yes to Substitue VIA Addr. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
In your second option using a STUN server would I need to setup my own STUN server? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Saturday, April 09, 2005 12:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] SPA and NAT traversal Thank you for your reply. There is a wealth of information on the wiki, etc. I turned on RTP debug and the SPA is not sending it's public IP it is sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere... Do I understand your question correctly: You have an SPA behind NAT1 and * and a second SIP device behind NAT2. Both devices register, but calls between the devices result in no audio? If that is the case, you can do one of two things: - set canreinvite=no for the devices' sip.conf entries, or - teach both devices to *stop* using their internal IPs for all communications and remove nat=yes from the entry for the SIP device inside NAT2. To set the SPA to give the correct IP, enable STUN, add a STUN server, and say Yes to Substitue VIA Addr. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA and NAT traversal
In your second option using a STUN server would I need to setup my own STUN server? No, use FWD or xten's STUN servers. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA and NAT traversal
The only way you ll be able to call extension to extension is if Asterisk is on the same node behind the nat. like the extensions or if each extension is on a different node. I run a proxie server and have ran through this problem many time. I bet you can call out bound to the outside world just fine from every extension. . Eric Wieling wrote: Jim Sturtevant wrote: I was hoping someone might help me diagnose a NAT issue with an SPA-2000 and my * server. My SPA is behind a NAT accessing a server which is also behind a NAT but SIP and RTP ports are forwarded to it. My SPA can successfully register. It can call another extension which is inside the * local net and the inside phone can call the SPA. But, no speech path either way. I have NAT=YES and the two invite parameters are set to NO. I'm desperately trying to get your sip.conf file telepathically but all I'm getting is images from your Martha Stewart porn collection. *shudder* In addition to nat=yes you also need localnet= and externip= set, as shown in sip.conf.sample. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
