I have this configuration:
UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet --> UA2
UA has provate address (192.168.x.x)
Asterisk has public address
I want to be reach somebody at the internet.
My idea was that asterisk works as a Proxy.
Then i would have a SIP/RTP connection between UA1 and Asterisk and an
other SIP/RTP connection between Asterisk and UA2. (asterisk is bridgeing
these calls). This setup solves the fu....ing NAT problem.
Question 1:
Is this possible ???
Question 2:
How do I have to configure extension.conf to reach somebody ??
what I want:
my UA1 tries to reach sip:test@example.com
my config:
exten => .,1,Dial(SIP/${EXTEN}) // whatever you get....try to
establish a SIP-session ......
but this expands to Dial(SIP/test) !!! Domain is gone !!!!
Question 3:
another idea. I also installed a SIP-Proxy. Then the chain is different:
UA1 ---> FW1(NAT) ---> Asterisk ----> SER (SIP-Proxy) --> FW2 -->
Internet
--> UA2
I still have the same problem: asterisk always tries to make a connection
to test@sip-proxy not to test@example.com
Questions (and Solutions):
A.
am i right that asterisk throws away the domain ??
if yes:
why don?t we have a environment variable with the Domain ??
example:
exten => .,1,Dial(SIP/${EXTEN}@${DOMAIN}) // whatever you get....try
to establish a SIP-session ......
B.
am i right that asterisk does not have the code to use a poxy ??
if yes:
why don?t we create some code to do it ?
example: (sip.conf)
[proxy]
type=proxy
host=10.10.10.10
and
exten => .,1,Dial(SIP/proxy) // whatever you get....try to establish
a SIP-session ......
C.
am i right that asterisk doesn?t support RFC3263 ( Session Initiation
Protocol (SIP): Locating SIP Servers)
if yes:
wouldn?t it be nice to write the code ???
any input is welcome........
Well can't help you with everything but at least i noticed your extension is incorrect ----- Original Message ----- From: "J?rg Bauer/Denic" <bauer@denic.de> To: <asterisk-users@lists.digium.com> Sent: Tuesday, April 22, 2003 5:19 PM Subject: [Asterisk-Users] howto> my config: > exten => .,1,Dial(SIP/${EXTEN}) // whatever you get....try to > establish a SIP-session ...... > > but this expands to Dial(SIP/test) !!! Domain is gone !!!!This is because your extension is missing the _ which tells Asterisk to look for a numer pattern. so replace it with this and it should work: exten => _.,1,Dial(SIP/${EXTEN}) I also asked (but haven't had any answer so far) if it would be possiible to use the VIA SIP-header. SIPphone <=> Asterisk <=> Firewall <=> SIPphone (or another SIPservice for that matter) It is very easy to map 5060 UPD (and TCP) to the Asterisk box, together with a range off for example 100 ports of RTP (8000-8100). After that the sip.conf might have something like: rtprange=800-8100 outsidehost=firewall.ourdomain.com That way Asterisk can add the outsidehost in the VIA header and the other side knows where to send the packets. I have send an email about this before to the list but i have know idea how can help me develop these settings into a patch for asterisk. If anyone would like to help be, please contact me (off list) and i'll get you some example SIP packets that support this option (Xlite phone, PingTel etc). Greetings, Tjardick
attached patch for rtpports= in sip.cfg and mgcp.cfg asterisk-users-admin@lists.digium.com wrote on 22.04.2003 20:54:21:> > It is very easy to map 5060 UPD (and TCP) to the Asterisk box, togetherwith> a range off for example 100 ports of RTP (8000-8100). > > After that the sip.conf might have something like: > > rtprange=800-8100renamed to: rtpports=8000-8123 -------------- next part -------------- A non-text attachment was scrubbed... Name: rtpports.patch.gz Type: application/octet-stream Size: 1526 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030423/a195e024/rtpports.patch.obj