One more thing may be useful for you. If you will get an error with cseq numder when provider send 401/407 message- usedialog module. It resole an issuevwith cseq( read documentation)
30.04.2015 18:23 пользователь "SamyGo" <govoiper@gmail.com> написал:
>
> I'd like you to google around, there is a function available from another module which will apply the changes in SIP Message.
>
>
> On Thu, Apr 30, 2015 at 9:51 AM, Ali Jibran <alijibran@vividtech.io> wrote:
>>
>> Perfect. Yeah got the working.
>>
>> Just one last issue. I don’t think this is rewriting the header. When I log the header again after the changes it still shows me the old values.
>>
>>
>>
>> From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of SamyGo
>> Sent: Thursday, April 30, 2015 6:50 PM
>>
>>
>> To: Kamailio (SER) - Users Mailing List
>> Subject: Re: [SR-Users] UAC Module
>>
>>
>>
>> t_on_failure("F_VOIP") to be used before t_relay();
>>
>> That will arm the call to go to F_VOIP on failure responses.
>>
>>
>>
>> On Thu, Apr 30, 2015 at 9:33 AM, Ali Jibran <alijibran@vividtech.io> wrote:
>>>
>>>
>>>
>>> #!ifdef WITH_FREESWITCH
>>>
>>> if(is_method("INVITE") && route(FROMFREESWITCH))) {
>>>
>>> xlog("L_INFO" ,"[$fU/$tU@$si:$sp]{$rm} Call from FreeSWITCH needs to be sent TOVOIP \n");
>>>
>>> route(TOVOIP);
>>>
>>> t_on_failure("F_VOIP");
>>>
>>> exit;
>>>
>>> }
>>>
>>>
>>>
>>> #!endif
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> route[TOVOIP] {
>>>
>>> xlog("L_INFO","ALERT: $fu to $tu ");
>>>
>>> $fU="XXXXXX";
>>>
>>> $td="sip.voipfone.net";
>>>
>>> $du="sip:XXXXXXX@sip.voipfone.net";
>>>
>>> t_relay();
>>>
>>>
>>>
>>> }
>>>
>>>
>>>
>>>
>>>
>>> failure_route[F_VOIP] {
>>>
>>> uac_auth();
>>>
>>> xlog("L_INFO","ALERT: IN FAIL");
>>>
>>> }
>>>
>>>
>>>
>>>
>>>
>>> I tried this but it never makes it to the failure branch. Im a newbie to kamailio and still working around the scripting. Can you please help me out here to where I am making the mistake?
>>>
>>>
>>>
>>> From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of SamyGo
>>> Sent: Thursday, April 30, 2015 9:18 AM
>>> To: Kamailio (SER) - Users Mailing List
>>> Subject: Re: [SR-Users] UAC Module
>>>
>>>
>>>
>>> Hi Jibran,
>>>
>>>
>>>
>>> Here is an old thread as reference:
>>>
>>>
>>> http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
>>>
>>>
>>>
>>> I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with username/password on a Provider for huge number of calls..imagine sending thousands of call to that provider and for each call going through the trouble of exchanging authentication.
>>>
>>> Thats why its usually recommended to go with IP-Authentication only. Send INVITE and Provider says Lets do this call,simple and easy.
>>>
>>>
>>>
>>> From the configuration perspective this is my idea of still using UAC.
>>>
>>>
>>>
>>> - Call coming from FS on kamailio
>>>
>>> - Rewrite the from-uri (so the provider receives calls from the registered username)
>>>
>>> - modify the to-domain part to contain the IP address of the provider
>>>
>>> - set the $du to ip of the provider, and t_relay() the call.
>>>
>>> - Most likely the Provider would say Proxy-Auth required..that can be caught in failure_route[]
>>>
>>> - There you can call the uac_auth() function to have username.password attached to the response of above. http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth()
>>>
>>> - once this function is successful send the INVITE again to the provider.
>>>
>>>
>>>
>>> Last three steps can be the following snippet of code(reference from here):
>>>
>>>
>>>
>>> failure_route[2] {
>>>
>>> if (t_check_status("40[17]")) {
>>>
>>> xlog("got challenged \n");
>>>
>>> if (uac_auth()) {
>>>
>>> xlog("auth was succesful \n");
>>>
>>> t_relay("udp:ip_addr:5060"); //provider's IP_ADDR
>>>
>>> }
>>>
>>> }
>>>
>>>
>>>
>>>
>>>
>>> I hope you get IP Auth from the provider, and find the reply useful.
>>>
>>>
>>>
>>> Regards,
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Apr 29, 2015 at 4:49 PM, Ali Jibran <alijibran@vividtech.io> wrote:
>>>>
>>>>
>>>> Hi all.
>>>> I have this setup.
>>>> Trunk--->Kamailio---->FreeSWITCH
>>>>
>>>> I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too.
>>>>
>>>> I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk?
>>>>
>>>> Basically this is what I'm trying to workout
>>>> FS---->kamailio---->trunk.
>>>>
>>>>
>>>> Any help will be much appreciated. Thanks.
>>>> AJ
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users@lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>>
>>>
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>>>
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>>
>>
>>
>>
>>
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>>
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>>
>>
>> _______________________________________________
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>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
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