Sip URI

kenewto

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I just upgraded to the newest version. Incredible PBX 3.

D-E-M-O doesn't connect to anything and I can't seem to connect to any SIP URI address. I thought it was native in PIAF, maybe a tutorial or something on it?


PIAF Installed Version = 2.0.6.2 Running on *HARDWARE*
FreePBX Version = 2.9.0.7
Running Asterisk Version = 1.8.8.0
Asterisk Source Version = 1.8.8.0
Dahdi Source Version = 2.6.0+2.6.0
Libpri Source Version = 1.4.12
Operating System = CentOS release 6.2 (Final)
Kernel Version = 2.6.32-220.2.1.el6.i686 - 32 Bit
Incredible PBX 3 Version = 3.0.3
 
D-E-M-O doesn't connect to anything and I can't seem to connect to any SIP URI address. I thought it was native in PIAF, maybe a tutorial or something on it?

Ward shut off 3366 (SIP URI) due to security concerns. How are you dialing the SIP address?
 
I am assuming you are dialing using a softphone. Can you please provide some logs. If I recall correctly, the functionality to dial SIP URIs from an endpoint isn't supported directly. The only way that I know that it works is if you add code in extensions_custom.conf to do the dialing manually.
 
I can't seem to find any useful Log files.


I'm looking at admin>Tools>Support>Asterisk Logfiles
 
Try running 'asterisk -vr' from SSH while you make the call. Post what you see, sanitized of course, here.
 
Nothing shows on the log. I can see the log reflecting other calls but when I send to a SIP URI, it doesn't show anything. Call displays shows "Trying" and I can hear ring back tone but it fails.
 
Does the domain (in your example, sipaddress.com) have the requisite SRV records published?

For example, if this is your own domain, you need an SRV record in DNS of _sip._udp.sipaddress.com pointing to 5060 pbx.sipaddress.com and your server expects communication over UDP on port 5060.

Code:
_sip._udp.sipaddress.com.    600    IN    SRV    0 0 5060 pbx.sipaddress.com.
If this isn't your domain, you can check to see if the record is published by using the dig utility. dig should already be installed on your PBX so you can run it from there.
Code:
dig srv _sip._udp.sipaddress.com.
This should return the something like the line above in the first example.
 
There is something to the DNS or domain that I am missing. I sent a call to SIP:[email protected] and it rang my extention. I've been thinking dialing rules but I think I am fighting a DNS issue.

SO... I am trying to call SIP:[email protected]. I immediatly get an error. Before I thought I was getting ringing but now I have discovered it wasn't.
 

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