[asterisk-users] SER + Asterisk

SIP sip at arcdiv.com
Tue Oct 21 05:03:00 CDT 2008


That's not actually true.  SER is very much alive and well and under 
constant development.

How do I KNOW it's constant development (other than the chatter on the 
mailing list)? Because things keep changing in CVS, but there never 
seems to be a 'release' version.  Just a release candidate. ;)

Seriously, though... this seems to be a popular misconception. I hear it 
a lot. Where did you come across the information that SER is no longer 
developed?

N.

Alex Balashov wrote:
> No, the issue isn't my value or preference.  The issue is that SER is no 
> longer maintained or developed and has not been for several years.
>
> Tobias Wolf wrote:
>
>   
>> Alex Balashov schrieb:
>>     
>>> SER is defunct.  Kamailio / OpenSIPS (formerly OpenSER) is the thing to do.
>>>   
>>>       
>> Well, i am not getting the correct meaning of 'defunct', but from the 
>> last part of your suggestion i guess you value Kamailio/OpenSIPS more 
>> than SER.
>>
>> Are there some hard reasion for this.
>>
>> I am in the process of deciding which SIP server i want to use with 
>> Asterisk and just made a go at SER. Compilation was a little rough but 
>> it was manageable. I threw away every module which funtionality i didn't 
>> wanted at after it just worked.
>>
>> I was able to register SIP phones at the server and configure an 
>> outgoing rule so that every call that could not be handled by the SIP 
>> server would go to Asterisk.
>>
>> But i confess, that i didn't looked at the other two projects ... Maybe 
>> they are so much better.
>>
>> Can you please write one or two aspects that makes me understand better 
>> why this two projects are the better choice ?
>>
>> Thank you very much ...
>>
>> Tobias
>>     
>>> On Fri, October 17, 2008 9:36 pm, Joseph wrote:
>>>
>>>   
>>>       
>>>> I am running Asterisk and would like to add SER to register my (sip) DID
>>>> and connect it to asterisk;
>>>> but I'm not sure if this is the correct forum.
>>>>
>>>> I have as DID, sip account with one VoIP provider; currently I"m using
>>>> just stand alone SIP phone and register with the VoIP provider via:
>>>> stun.fwdnet.net
>>>>
>>>> Is it possible to use SER to register with the provider and forward the
>>>> call Asterisk.
>>>> Can anybody provide a link to practical example.
>>>>
>>>> I'm comfortable with Asterisk but I just install SER and can not find
>>>> appropriate example to follow on "www.iptel.org" web-page.
>>>> There are a lot explanations but not enough practical examples to follow.
>>>>
>>>> --
>>>> #Joseph
>>>>
>>>> _______________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>>     
>>>>         
>>>   
>>>       
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>     
>
>
>   




More information about the asterisk-users mailing list