[Serusers] SER + Asterisk

[Laurent BURGY]

I answer to myself , i forgot a break; after the rewritehostport ...
Sorry,
Laurent

Laurent BURGY wrote:

> Hi,
> First of all, thanks a lot Alessio for your help ... I m not so far 
> away from the end of my quest....
> I m still encountering one problem.... When i pass call to the pstn 
> from ser through asterisk, even if it succeeds, the value on my client 
> is still 404... My mobile phone is ringing but i see on the Xlite 
> screen 404, i don't understand ...
>
> thx
> Laurent
> Alessio Focardi wrote:
>
>> Hello Laurent,
>>
>> Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
>>
>> LB> I think i follow the instructions ... but should i "route"  ( so 
>> modify
>> LB> the ser.cfg file)  some messages to my gateway to use the answering
>> LB> machine or services provided by my gateway ...
>> LB> And what about autocreatepeer=yes?
>>
>> surely later you will have to route in ser.cfg for outbound calls, 
>> checking first for
>> user auth and grp assignement.
>>
>> Using the extention method you can direct sip addresses to services in
>> asterisk like voicemail or voice prompts.
>>
>> that's my ser routing for outside calls
>>
>>
>>
>>                                        
>>                                        
>> record_route();                 
>>                                                              
>>                                        if (uri=~"sip:0[0-9]+@"){
>>                                                                       
>> if (!proxy_authorize("mydomain.com", "subscriber")) 
>> {proxy_challenge("mydomain.com", "0");sl_send_reply("403", "That's 
>> not your home");break;}; #fine proxy challenge
>>                                        if (!is_user_in("credentials", 
>> "local")){sl_send_reply("403", "No permission for local 
>> calls");break;}; #fine invite                               
>>                                                       
>> rewritehostport("sip.mydomain.com:5090");
>>                                        t_relay();
>>                                        break;
>>                                                                       
>> }; #fine if uri sip:0
>>
>> inside asterisk calls forwarded by ser are treated by this extention
>>
>> exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
>> exten => _0.,2,Congestion
>>
>> its important to notice that you have to block port 5090 for incoming
>> ip requests ....
>>
>> Only ser will be allowed to talk to asterisk and forward calls, after
>> user checking.
>>
>> Hope it helps
>>
>> LB> thx
>>
>> LB> Alessio Focardi wrote:
>>
>>  
>>
>>>> Hello Laurent,
>>>>
>>>> Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
>>>>
>>>> LB> But my clients should register on SER or on Asterisk?
>>>>
>>>> On ser, then you will need to protect asterisk from unallowed pstn
>>>> call, but that will come later on.
>>>>
>>>>
>>>> LB> thx
>>>>
>>>> LB> Alessio Focardi wrote:
>>>>
>>>>
>>>>
>>>>     
>>>>
>>>>>> Hello Laurent,
>>>>>>
>>>>>> Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>>>>>
>>>>>> LB> In fact, my ser installation works fine...
>>>>>> LB> I can pass call through asterisk in standalone...
>>>>>> LB> The problem is to interconnect the 2, to register ( i don't 
>>>>>> know if it's
>>>>>> LB> a right solution) the sipphones on SER and to go outside 
>>>>>> thanks to
>>>>>> LB> asterisk...
>>>>>>
>>>>>> make asterisk use port 5090 for sip, then as a first step make
>>>>>> asterisk register in ser as an extention.
>>>>>>
>>>>>> you can do this in asterisk's sip.conf
>>>>>>
>>>>>> example
>>>>>>
>>>>>> register => 10:password at sip.yourdomanin.com/10
>>>>>>
>>>>>> this tells asterisk to register extention 10 as address 
>>>>>> 109 at yourdomain.com
>>>>>>
>>>>>> dial 10 with a sip phone and you are in asterisk ... note that you
>>>>>> should have an extention 10 defined, or it will not work.
>>>>>>
>>>>>> B> Thx
>>>>>>
>>>>>> LB> Alessio Focardi wrote:
>>>>>>
>>>>>>
>>>>>>
>>>>>>    
>>>>>>         
>>>>>>
>>>>>>>> Hello Laurent,
>>>>>>>>
>>>>>>>> What you want to accomplish could be done, my advice is to setup a
>>>>>>>> working installation of ser then you will continue with asterisk.
>>>>>>>>
>>>>>>>> At first make a simple installation of ser (no auth, no db 
>>>>>>>> maybe) and make
>>>>>>>> your phones call each other.
>>>>>>>>
>>>>>>>> If you encounter specific problems and you want to have some 
>>>>>>>> help this
>>>>>>>> is the right place.
>>>>>>>>
>>>>>>>> Good luck !
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>>>>>
>>>>>>>> LB> Hi,
>>>>>>>> LB> I want to use SER as a sip Proxy and asterisk as a gateway 
>>>>>>>> to the PSTN
>>>>>>>> LB> network ...
>>>>>>>> LB> My sipphones are BudgetTone101 and i'm having trouble 
>>>>>>>> trying configure
>>>>>>>> LB> them....
>>>>>>>> LB> Indeed, i don't know if they should register on SER or 
>>>>>>>> not... I don't
>>>>>>>> LB> know what kind of sip messages should be passed to my 
>>>>>>>> machine running
>>>>>>>> LB> asterisk.
>>>>>>>> LB>    I don't know what must be in ser.cfg ( if you've an 
>>>>>>>> example it could
>>>>>>>> LB> help me a lot...)...
>>>>>>>> LB> I wasn't able to find documentations about using Ser and 
>>>>>>>> Asterisk in
>>>>>>>> LB> this configuration ( messages in the archives are not explicit
>>>>>>>> LB> enough....) , so if you've a pointer or so....
>>>>>>>>
>>>>>>>>
>>>>>>>> LB>    Help...
>>>>>>>>
>>>>>>>> LB> thx,
>>>>>>>> LB> Laurent
>>>>>>>>
>>>>>>>> LB> _______________________________________________
>>>>>>>> LB> Serusers mailing list
>>>>>>>> LB> serusers at lists.iptel.org
>>>>>>>> LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>   
>>>>>>>>        
>>>>>>>>             
>>>>>>>
>>>>>> LB> _______________________________________________
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>>>>>> LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>    
>>>>>>         
>>>>>
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>>>>
>>>>     
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