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[asterisk-users] SIP show peers: UNREACHABLE


 
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hawat.thufir at gmail.com
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PostPosted: Mon Mar 16, 2015 8:14 pm    Post subject: [asterisk-users] SIP show peers: UNREACHABLE Reply with quote

Page 176 of Asterisk, the definitive manual, discusses "Connecting an Asterisk system to a SIP provider" in the context of, at least the concept of, "trunking".

Previously, I wasn't able to connect to the peer, and attributed that to a combination of double NAT (plus), and latency and lag due to wi-fi.  Now that I'm directly connected to the cable modem (well, gateway router and modem combo), the connection is better and I'm able to make outgoing VoIP calls with Jitsi.

Am I right in thinking that the very same connection parameters I entered in Jitsi will work fine when entered in Asterisk with syntax like:

register => username:password@your.provider.tld ([email]username:password@your.provider.tld[/email])

and by creating the peer entry in sip.conf for the service provider.

One concern is that the provider uses:

  1. User ID can be any one of your 11-digit babyTEL telephone numbers. Typically your main number but can be any one of your other phone numbers.
  2. For your protection the SIP Password field does not reveal your password until you explicitly click on ‘Show password’.
  3. If Outbound Proxy is not supported on your system, try one of the following two options:
    1. Add the line “198.38.7.34 sip.babytel.ca” to your system’s “hosts” file and configure the SIP Proxy as: “sip.babytel.ca:5065”. This uses the TCP/IP “hosts” file address mapping mechanism to redirect SIP traffic to the Outbound Proxy.
    2. Configure the SIP Proxy as: “198.38.7.34:5065”. This replaces the SIP Proxy address with a resolved Outbound Proxy address.


On a mac, I added that line to the hosts file -- but I'm not sure it's required.  How do I specify the SIP proxy and the outbound proxy?  What's the distinction between a SIP proxy and outbound proxy?



In Jitsi, I configured as 123456789@sip.babytel.ca (123456789@sip.babytel.ca) for SIP id.

In "Connection" I used "sip.babytel.ca" for the registrar and the user, 1234567890, as the the authorization name.  I put the proxy as nat5.babytel.ca, port 5065 and the preferred transport as UDP.  I don't see all those options, particularly surrounding the proxy and outbound proxy.  Again, I'm unclear on why there's a proxy specificed, and then a different outbound proxy is specified as well.




How do I establish that I've entered the parameters correctly in Asterisk?  Or, how do I establish that the parameters are incorrectly entered?  Because Jitsi is able to call out and in, I believe that eliminates NAT, firewall or other networking issues. 



thanks,

Thufir
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