[Freeswitch-users] context

Steven Ayre steveayre at gmail.com
Thu Feb 3 14:46:45 MSK 2011 

Everything in the 1.0.6 book should work on 1.0.7 though...

It'll only be missing some of the new features, but they're on the Wiki

-Steve


On 3 February 2011 11:46, Steven Ayre <steveayre at gmail.com> wrote:

> No, and it never will be... 1.0.7 is the daily snapshot release candidate
> for the 1.0.8 release. The book might be updated for 1.0.8 once it comes out
> though.
>
> -Steve
>
>
> On 3 February 2011 11:24, Sam <u2nsam at gmail.com> wrote:
>
>> Is FS book 1.0.7 available ?
>>
>> Regds
>>
>>
>>
>> On Thu, Feb 3, 2011 at 4:37 PM, Sam <u2nsam at gmail.com> wrote:
>>
>>> Wow,
>>>
>>> super explanation of the call flow, if such explanations are there in the
>>> FS book , I will take it.
>>>
>>> Thanks again.
>>>
>>> Regds
>>> Sam
>>>
>>>
>>> On Thu, Feb 3, 2011 at 12:15 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> I think I'm going to write an article on this topic for the new folks
>>>> since it seems to be a common point of confusion. Don't worry, you're not
>>>> alone - this is a common challenge for new FS admins.
>>>>
>>>> #1 - The first line of your pastebin is the key. You are letting the
>>>> call in via an ACL. This is "okay" if it is what you want, but I doubt that
>>>> it is. I recommend that you remove this phone's IP address from the
>>>> "domains" section of acl.conf.xml. When using the default FS configs, if you
>>>> let a phone call through using the "domains" ACL then it automatically goes
>>>> to the public context. Why? Because the call is not explicitly associated
>>>> with a local user on the system. It's like getting a free pass into
>>>> Disneyland but you don't have a little badge that says, "Hi, my name is..."
>>>>
>>>> Some get confused by the "Falling back to digest auth" message. It looks
>>>> like an error but it's really just information. If the caller's SIP client
>>>> properly authenticates (which is different than registration - see below)
>>>> then FS knows exactly who is making the call and that it is a local user so
>>>> it goes into the "default" context.
>>>>
>>>> #2 - Make sure that you learn the difference between SIP *registration*
>>>> and SIP *authentication*.  Registration is where the SIP client tells FS:
>>>> "Here's how you can reach me if you get a call for me." It is for calls TO
>>>> the phone, not calls from the phone. On the other hand, call authentication
>>>> is for calls FROM the phone to FS. When the phone calls FS, FS first checks
>>>> to see if the phone's IP is in the "domains" ACL. If it is, then it just
>>>> let's the call in to be handled by the public context. If not, then FS sends
>>>> an "auth challenge" - basically saying, "What's the password?" If the SIP
>>>> client properly responds then the call is "authenticated" - meaning that FS
>>>> knows it is from a specific user and thus it goes to the default context.
>>>>
>>>> I recommend that you get the FS book and look at chapter 4. We talk a
>>>> lot about the user directory and it will help you understand how it all
>>>> works. Hopefully this explanation will whet your appetite for more. :)
>>>>
>>>> Keep hacking away at it - you'll get it soon enough!
>>>>
>>>> -MC
>>>>
>>>> On Wed, Feb 2, 2011 at 7:02 PM, Sam <u2nsam at gmail.com> wrote:
>>>>
>>>>> So you say that the extensions are not registered with FS and they are
>>>>> getting allowed by the acl, just like DIDs.
>>>>> How can i rectify this,any thing that needs to be checked ?
>>>>>
>>>>> http://pastebin.freeswitch.org/15217
>>>>>
>>>>> Regards
>>>>> Sam
>>>>>
>>>>>
>>>>> On Wed, Feb 2, 2011 at 11:23 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Turn on console debug level output (default in fs_cli) and make the
>>>>>> test call. Pastebin the output. Most likely the call is not being authorized
>>>>>> because you are letting it in via an ACL or something like that.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>> On Wed, Feb 2, 2011 at 1:55 AM, Sam <u2nsam at gmail.com> wrote:
>>>>>>
>>>>>>> I have defined a user as :
>>>>>>>
>>>>>>> <user id="2099">
>>>>>>>       <params>
>>>>>>>         <param name="password" value="1234"/>
>>>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>> ${dialed_domain})}"/>
>>>>>>>         <param name="mwi-account" value="2099 at voicemail"/>
>>>>>>>         <param name="vm-password" value="1234"/>
>>>>>>>         <param name="vm-email-all-messages" value="true"/>
>>>>>>>         <!--param name="vm-attach-file" value="true"/-->
>>>>>>>         <!--param name="vm-mailto"  value=""/-->
>>>>>>>       </params>
>>>>>>>       <variables>
>>>>>>>         <variable name="user_context" value="inter"/>
>>>>>>>         <variable name="force_transfer_context" value="inter"/>
>>>>>>>         <variable name="accountcode" value="2099"/>
>>>>>>>         <variable name="internal_caller_id_name" value="2099"/>
>>>>>>>         <variable name="internal_caller_id_number" value="2099"/>
>>>>>>>         <variable name="callgroup" value="1"/>
>>>>>>>         <variable name="sip-force-contact"
>>>>>>> value="NDLB-connectile-dysfunction"/>
>>>>>>>       </variables>
>>>>>>>
>>>>>>>
>>>>>>> and in the dialplan i have
>>>>>>>
>>>>>>> <context name="inter">
>>>>>>>
>>>>>>> <extension name="2099">
>>>>>>>     <condition field="destination_number" expression="^(2099)">
>>>>>>>       <action application="set" data="call_timeout=30"/>
>>>>>>>       <action application="set" data="dialed_extension=$1"/>
>>>>>>>             <action application="export" data="dialed_extension=$1"/>
>>>>>>>             <action application="set"
>>>>>>> data="transfer_ringback=$${hold_music}"/>
>>>>>>>             <action application="set"
>>>>>>> data="hangup_after_bridge=true"/>
>>>>>>>             <action application="set" data="continue_on_fail=true"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
>>>>>>>             <action application="set"
>>>>>>> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>>>>>> var callgroup)}"/>
>>>>>>>             <action application="hash"
>>>>>>> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
>>>>>>>             <action application="bridge"
>>>>>>> data="{sip_invite_domain=$${domain}}user/${dialed_extension}@
>>>>>>> ${domain_name}"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>> </context>
>>>>>>>
>>>>>>>
>>>>>>> Now extension 2075 is also made on above lines in the same context
>>>>>>> for user and dialplan, here the call when initiated from 2075 to 2099
>>>>>>> searches for context public instead of context inter
>>>>>>>
>>>>>>> mod_dialplan_xml.c:331 Processing 2075 <2075>->2099 in context public
>>>>>>>
>>>>>>> any reason why ?
>>>>>>>
>>>>>>> Regards
>>>>>>> Sam
>>>>>>>
>>>>>>> _______________________________________________
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>>>>>>>
>>>>>>>
>>>>>>
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>>>>>>
>>>>>
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