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-   -   SipBroker in Japanese press (https://forum.sipbroker.com/showthread.php?t=1499)

martin 05-15-2007 06:28 AM

Quote:

Originally Posted by kieranmullen (Post 8899)
I am surprised how quick the lookup is via the website. I wonder why it is sooo slow via asterisk. I am using freepbx. A fix was posted however I have no idea if it was ever implemented...
KM

We wrote our own lookup that kicks off multiple threads in parallel (one for each root). We actually wrote a web service that is available to developers. Maybe the asterisk developers should look at our service as a faster alternative ;)

SIPBroker Web Service Documentation
.

v164 06-23-2007 03:40 AM

dial a spiral
 
Quote:

Originally Posted by v164
SIP Broker has been given the prefix "0101", and voxalot.com has been given the entry "0100".

(Conversely, SIP IX has been assigned the prefix "*8096" in SIP Broker)

That means you can dial what RFC 3261 refers to as a "spiral" (as opposed to a "loop").

Eg, to dial Voxalot number 123456, you could dial:


Code:

*809601018829993123456 @ sipbroker.com
which would be converted to

Code:

    01018829993123456 @ sipix.jp
which would be converted to

Code:

        8829993123456 @ sipbroker.com
which would be converted via ENUM (e164.org) to

Code:

                123456 @ voxalot.com

For this call, the SIP INVITE message that arrives from 64.34.173.199:5060 (us.voxalot.com) is this:

(Voxalot number is call forwarded to sip uri: sip:050xxxxxxxx@xxx.xxx.xxx.xxx:5060)

Code:

INVITE sip:050xxxxxxxx@xxx.xxx.xxx.xxx:5060 SIP/2.0
Record-Route: <sip:64.34.173.199;lr=on;ftag=2568014107>
Record-Route: <sip:64.34.162.221;lr=on;ftag=2568014107>
Record-Route: <sip:64.34.162.221;lr=on;ftag=2568014107>
Via: SIP/2.0/UDP 64.34.173.199;branch=z9hG4bKd728.14ebdfe4.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKd728.bbb7b4e1.0
Via: SIP/2.0/UDP 58.158.169.93;branch=z9hG4bKd728.db54b947.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKd728.abb7b4e1.0
Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:6080;branch=z9hG4bK24374553867812144;rport=6080
From: 61xxxxxxxxx <sip:61xxxxxxxxx@sip.pennytel.com>;tag=2568014107
To: "*809601018829993123456" <sip:*809601018829993123456@sipbroker.com:5060>
Call-ID: 238426250-49112365226@aaa.bbb.ccc.ddd
CSeq: 1 INVITE
Contact: <sip:61xxxxxxxxx@aaa.bbb.ccc.ddd:6080>
max-forwards: 14
supported: replaces, 100rel
user-agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 342
P-hint: call forward

Reading the "Via:" headers from bottom to top, it can be seen that, as expected, the SIP INVITE message has traversed, in this order:

aaa.bbb.ccc.ddd (the SIP device originating the call)

64.34.162.221 ( sipbroker.com )

58.158.169.93 ( sipix.jp )

64.34.162.221 ( sipbroker.com )

64.34.173.199 ( us.voxalot.com )


(It's interesting to note that sipbroker.com and us.voxalot.com both add the optional "Record-Route:" header, whereas sipix.jp does not).


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