[VoIP] Is Office code 688 sent up right?

john jones jjones3601 at yahoo.com
Sun Oct 28 20:34:02 CST 2007


So it sounds like the way Greg has ENUM configured is correct.  But what would  the sip.conf  entry look like?  It seems that it allows guests by default and allows externalinvites by default as well.  In my system, this should come into the extensions processing in the default context.

Do I have this concept right?

Thanks!

John




[general]                                                                                                                
context=default                 ; Default context for incoming calls                                                     
;allowguest=no                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'             
                                ; if asterisk was compiled with OSP support.                                             
;realm=mydomain.tld             ; Realm for digest authentication                                                        
                                ; defaults to "asterisk"                                                                 
                                ; Realms MUST be globally unique according to RFC 3261                                   
                                ; Set this to your host name or domain name                                              
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)                                        
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)                                           
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls                                               
                                ; Note: Asterisk only uses the first host                                                
                                ; in SRV records                                                                         
                                ; ability to place SIP calls based on domain                                             
                                ; names to some other SIP users on the Internet                                          
                                                                                                                         
;domain=mydomain.tld            ; Set default domain for this host                                                       
                                ; If configured, Asterisk will only allow                                                
                                ; INVITE and REFER to non-local domains                                                  
                                ; Use "sip show domains" to list local domains                                           
;domain=mydomain.tld,mydomain-incoming                                                                                   
                                ; Add domain and configure incoming context                                              
                                ; for external calls to this domain                                                      
;domain=1.2.3.4                 ; Add IP address as local domain                                                         
                                ; You can have several "domain" settings                                                 
;allowexternalinvites=no        ; Disable INVITE and REFER to non-local domains                                          
                                ; Default is yes                                                                         
;autodomain=yes                 ; Turn this on to have Asterisk add local host                                           
                                ; name and local IP to domain list.                                                      
;pedantic=yes                   ; Enable slow, pedantic checking for Pingtel                                             
                                ; and multiline formatted headers for strict                                             
                                ; SIP compatibility (defaults to "no")                                                   
;tos=184                        ; Set IP QoS to either a keyword or numeric val                                          
1                                                                                         

----- Original Message ----
From: Shane Young <voiptandem at shaneyoung.com>
To: voip at ckts.info
Sent: Sunday, October 28, 2007 9:38:06 PM
Subject: Re: [VoIP] Is Office code 688 sent up right?


There is no concept of "context" within SIP itself and therefore you  
wouldn't see it in an ENUM entry.

Context, as we use it in this discussion, is  an Asterisk specific  
thing and is why you can include it in the "Inter-Asterisk eXchange  
version 2 protocol (IAX2)" ENUM entry.

In Asterisk configuration, you specify on the inbound peer what  
context to use.  IAX2 gives you the ability to specify multiple  
contexts for that peer and alllow the peer to chose which context they
  
want.  The originating asterisk system can specify what context they  
want to use in the call setup.

I don't know if that helps explain it or not.

--Shane



Quoting Greg Blakely <greg at vyger.net>:

> John,
>
> I am not an expert on SIP entries in ENUM.  So, at this time, let me
 ask
> the group for help on this matter. One of the things I'm unclear
 about
> is how to include a context entry in SIP connections.  Hopefully,
> someone has a leg up on this, and can get me headed down the right
 road.
>
> First, though, \\1 translates roughly to "the digits that were
 queried,"
> usually the digits that were dialed.  So, when I dial 16887070, the
 enum
> entry I have for you should have me dialing
 SIP:16887070 at sxs.jj3601.com.
>
>
> As you can see, it includes NO incoming context.
>
> Looking at how I dial out through iConnectHere, a SIP provider, I
 just
> send the connection request to number at theirserver.  I authenticate by
> registering with them -- which is completely separate from an ENUM
> lookup or dialing string.
>
> I have **got** to be wrong about this!  At least, I hope I am.
>
> Anyone??????
>
>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info]
>> On Behalf Of john jones
>> Sent: Sunday, October 28, 2007 6:57 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: [VoIP] Is Office code 688 sent up right?
>>
>> Greg,
>>
>> These look very different but maybe they are supposed to?  It
>> looks like maybe the incoming context for 688 is \\1 ???
>>
>>
>> Thanks!
>>
>> John
>>
>> root at Demo_OpenWRT_688:~# dig 1.8.8.6.1.std.ckts.info  NAPTR
>>
>> ;; ANSWER SECTION:
>> 1.8.8.6.1.std.ckts.info. 60     IN      NAPTR   100 10 "u" "E2U+SIP"
>>  "!^\\+*([^\\*]*)!sip:\\1 at sxs.jj3601.com!" .
>>
>> root at Demo_OpenWRT_688:~# dig 1.7.8.6.1.std.ckts.info  NAPTR
>>
>> ;; ANSWER SECTION:
>> 1.7.8.6.1.std.ckts.info. 60     IN      NAPTR   100 10 "u"
 "E2U+IAX2"
>>  "!^\\+*(.*)$!iax2:inbound at linksys.jj3601.com/\\1!"
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
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>

--Shane
+1-821-7311 CNET

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