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{{Box Important|This page was written in 2005 and is a bit out of date. Most of the reasons why you would want to interconnect sipX with Asterisk have gone away as sipX has not only caught up in terms of supported features, but in more and more areas has taken the lead, is more robust and easier to use. We expect therefore that this page will be deleted early 2007 after we finish sipX release 3.8.|} -Still here as of Oct. 2014. 

 

Another thing to note that as of Asterisk version 1.6 the refer method for transferring calls is not supported in sipX. Please refer to Asterisk documentation regarding SIP Refer method used by Asterisk. An untested work around since transfers worked in Asterisk version 1.4 is to use the option in sip.conf called pedantic, which will either make Asterisk pay more attention to SIP call-IDs or not. 

Ref http://www.voip-info.org/wiki/index.php?page_id=2768

pedantic = yes|no : Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. Default no (in Asterisk 1.8 default yes)

 

Basically, you can think of the pedantic setting as being one that is important
to SIP purists or those who do make use of more advanced/less used nuances of
the SIP protocol.

 

For instance, RFC 3261 plainly spells out that the way to identify whether an
incoming message belongs to a specific dialog is to check the call-id, to tag,
and from tag. With pedantic mode disabled, all we look at is the call-id because
in the majority of situations, the call-id is suitable for identifying which
dialog the request belongs to. However, if you know that tag checking is
important, then it will be enabled by turning on pedantic mode.

 

There are other RFC-mandated operations which will only occur if pedantic mode
is enabled. Typically, the reason why we restrict these operations to when
pedantic mode is enabled is because it is a lot easier/more efficient not to
perform the operations and because in most cases things work fine without
performing the operations.

 

Mark Michelson


Introduction - Interconnecting sipXecs IP PBX with the Asterisk PBX

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Panel

#Remove the hash (#) character in the line below (before the word ztdummy at the end of the line)

No Format
MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \{noformat}
 

ztdynamic ztd-eth wct1xxp wct4xxp # ztdummy

No Format
 

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The following set of commands compiles and installs both the zaptel drivers as well as Asterisk for a 2.6 Linux kernel (Fedora Core 3).
{{Box Code|Compile zaptel, libpri and Asterisk|

...

Panel
  1. 2XX calls the SIP extension of the domain directly (using DNS SRV call goes to sipX)
  2. 100 calls the auto-attendant on sipX
  3. 101 calls directly into sipX voicemail
    ;
    outgoing

    No Format
    exten => _2XX,1,Dial(Sip/${EXTEN}@your.domain,30,t)
    No Format
    exten => 100,1,Dial(Sip/${EXTEN}@your.domain,30,t)
    No Format
    exten => 101,1,Dial(Sip/${EXTEN}@your.domain,30,t)
    No Format
    exten => t,1,Hangup()

...