[asterisk-users] Re: Re: SIP "Lines" Example Citel

[Jason Fuermann]

We have done limited testing with the Citel gateways and they seem 
pretty cool. We're fixing to deploy them as a replacement to a hotel 
pbx, and after that use them as an interim solution until full VoIP 
convergence in our campus environment. I would be interested to know 
what other peoples experiences are with Citel

Chris Earle wrote:
> I'm in the middle of trying to solve that very problem -- integrate a legacy
> pbx into my asterisk/IP network.  (Norstar MICS for the record)
>
> Been mulling over how to do it with Quintum boxes the last few days --- very
> complicated.  The alternative solution is to use a Citel gateway with the
> 30+ handsets we have and scrap the MICS pbx and build an asterisk box with
> our incoming POTS lines into a digium/sangoma card.
>
> Any testimonials about these approaches? (Quintum boxes or Citel handset
> gateways)  Please share!
>
> --
> Chris Earle
> System Solutions Specialist
>
>
>
> "Steve Totaro" <stotaro at asteriskhelpdesk.com> wrote in message
> news:4511CA28.5060408 at asteriskhelpdesk.com...
>   
>> Steven wrote:
>>     
>>> Setting Realm to asterisk worked for me.
>>>
>>> ref. from sip.conf:
>>> ;realm=mydomain.tld  ; Realm for digest authentication
>>>     ; defaults to "asterisk". If you set a system name in
>>>     ; asterisk.conf, it defaults to that system name
>>>     ; Realms MUST be globally unique according to RFC 3261
>>>     ; Set this to your host name or domain name
>>>
>>>
>>>
>>>
>>>       
>> Yes it worked for me too.  Awesome box if you have to integrate/migrate
>> from a legacy system to Asterisk but keep your old handsets.  I am very
>> impressed.  The two devices I really like now are the Citel Gatways and
>> the Quintum boxes.
>>
>> Thanks,
>> Steve Totaro
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>
>
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