I'm trying to configure SIP trunking. Now, I'm referencing
"Asterisk
the definitive guide", 4th ed. While I don't have the page handy, I
was
reading the suggestion to try SIP to SIP before proceeding to outside
connectivity. I'm aware that SIP trunking is a construct, but am,
obviously, learning the system.
What I'd like to do is from the CLI "ping" either the peer below,
or a
peer somewhere. Unfortunately, I'm also in a double+ NAT situation at
the moment. While Skype works (mostly) from my LAN, the connection
isn't the greatest. My LAN uses a wireless bridge to connect to another
LAN. It's just a home setup; it is what it is.
How do I test a connection? How do check the settings? As far as I
can tell, the settings are correct.
tleilax:~ #
tleilax:~ # asterisk -V
Asterisk 1.8.32.1-vici
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
========================================================================log and
verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid =
3062)
Verbosity is at least 21
tleilax*CLI>
tleilax*CLI> sip show peer babytel
* Name : babytel
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Subscr.Cont. : <Not set>
Language : en
AMA flags : Unknown
Netborder CPD: No
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Yes
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : sip.babytel.ca
Addr->IP : 198.38.7.11:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1<private>
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : UNREACHABLE
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
tleilax*CLI>
tleilax*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201 (Unspecified) D N 0 UNKNOWN
babytel/1<private> 198.38.7.11 D
N 5060 UNREACHABLE
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
tleilax*CLI>
thanks,
Thufir
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 at your.provider.tld
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 at 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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