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" <govoi...@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 <alijib...@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-boun...@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 <alijib...@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:xxxx...@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-boun...@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 <alijib...@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 >>> >>> >>> >>> >>> >>> ________________________________ >>> >>> This email is free from viruses and malware because avast! Antivirus protection is active. >>> >>> >>> >>> >>> _______________________________________________ >>> 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 >> >> >> >> >> >> ________________________________ >> >> This email is free from viruses and malware because avast! Antivirus protection is active. >> >> >> >> _______________________________________________ >> 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 >> > > > _______________________________________________ > 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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