[asterisk-users] PJSIP outbound register and inbound calls

[Nick Awesome]

oh.. its simple.

"[res_pjsip_endpoint_identifier_ip]" should be before "identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]”

Thanks all for help :)

On 17 Jul 2014, at 11:05, Nick Awesome <jleed at me.com> wrote:

> New information, as I said I’m using realtime,
> thats the problem!
> 
> I just tested using static config file and it is working perfect.
> after some research I figured out that problem with “ps_endpoint_id_ips" for some reason asterisk ignoring matches in this table,
> 
> I have string in sorcery.conf
> 
> identify = realtime,ps_endpoint_id_ips
> 
> also have string in extconfig.conf
> 
> ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips
> 
> and ofc I have table
> 
> CREATE TABLE `pbx_endpoint_id_ips` (
>   `id` varchar(40) NOT NULL,
>   `endpoint` varchar(40) DEFAULT NULL,
>   `match` varchar(80) DEFAULT NULL,
>   UNIQUE KEY `id` (`id`),
>   KEY `ps_endpoint_id_ips_id` (`id`)
> ) ENGINE=InnoDB DEFAULT CHARSET=latin1;
> 
> with entry 
> 
> 10001 | 10001 | 85.195.98.178
> 
> but thats just didn’t works(
> 
> is this a bug and should I open ticket ?
> 
> On 16 Jul 2014, at 21:13, Nick Awesome <jleed at me.com> wrote:
> 
>> Ok there is my test account from sipiko.net
>> 
>> username: cb5069
>> password: sqv664yqtp
>> domain: callme.sipiko.net
>> 
>> its using username/password authentication.
>> because its just website widget I need only inbound calls from this peer,
>> test call can be done from url: 
>> http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes
>> 
>> on my side I have an asterisk 12 using pjsip
>> 
>> Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context
>> 
>> help if you can please:)
>> 
>> On Jul 16, 2014, at 8:53 PM, Joshua Colp <jcolp at digium.com> wrote:
>> 
>>> Nick Awesome wrote:
>>>> I thought that
>>>>>> type=identify
>>>> will match an IP address and accept it,
>>>> 
>>>> well, in my example I can control both sides and able to configure it
>>>> without registration. in real life I have a provider that requires
>>>> username/password authentication
>>>> 
>>>> provider gives me - Username - Password - DomainName
>>> 
>>> They may require it for *outgoing* calls to them but for incoming I
>>> highly doubt they'd want you to authenticate them. It's usually always
>>> IP authentication.
>>> 
>>>> I have configure it like I showed before and have exactly the same
>>>> notice
>>>> 
>>>> [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
>>>> log_unidentified_request: Request from
>>>> '"cb5069"<sip:asterisk at 85.195.98.178>' failed for
>>>> '85.195.98.178:5060' (callid:
>>>> 173995aa2e25283807700d65055c9214 at 85.195.98.178) - No matching
>>>> endpoint found 85.195.98.178 is an operator,
>>>> 
>>>> so what I should add to my config to be able accept calls from
>>>> Registered peer ?
>>> 
>>> The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.
>>> 
>>> Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.
>>> 
>>> -- 
>>> Joshua Colp
>>> Digium, Inc. | Senior Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & www.asterisk.org
>>> 
>>> -- 
>>> _____________________________________________________________________
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>> 
>> -- 
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>> 
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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