[OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

[Satish Patel]

I downloaded latest repo from git whatever you mentioned. At compile time I got following error when it was compiling uac.c 

Fatal error: auth.h file and directory not found. 

Does it require any dependencies? 

Sent from my iPhone

On Aug 25, 2014, at 8:10 AM, Vlad Paiu <vladpaiu at opensips.org> wrote:

> Hello,
> 
> Ok then, the patches should apply just fine - 
> https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
> 
> was already backported, so just apply 
> https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33
> 
> and test it out.
> 
> Best Regards,
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.com 
> On 25.08.2014 13:36, Satish Patel wrote:
>> I'm using 1.11 last week installed. 
>> 
>> Sent from my iPhone
>> 
>> On Aug 25, 2014, at 6:34 AM, Vlad Paiu <vladpaiu at opensips.org> wrote:
>> 
>>> Hello,
>>> 
>>> What OpenSIPS version are you currently using ?
>>> I've just committed a fix that implements a preliminary version of this, see commits :
>>> 
>>> https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
>>> and
>>> https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33
>>> 
>>> Please apply them to your sources and let me know how it oges
>>> Best Regards,
>>> Vlad Paiu
>>> OpenSIPS Developer
>>> http://www.opensips-solutions.com 
>>> On 25.08.2014 13:31, Satish Patel wrote:
>>>> Great!! I can see light in tunnel now because last 1 week I tried everything and now I was planing to go for B2B but I guess as you said you guys working on so I'm holding my breath. 
>>>> 
>>>> This is must needed solution because SIP service provide most of time provide password to make outbound trunk call. 
>>>> 
>>>> Sent from my iPhone
>>>> 
>>>> On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
>>>> 
>>>>> Hi Satish,
>>>>> 
>>>>> It is an known issue that OpenSIPS does not increases the cseq number when performing UAC auth against another party. Asterisk does not like that and consider the new branch INVITE with credentials a simple retransmission (even if it has a different VIA-branch :P) and discards them - this is why you get that timeout from asterisk.
>>>>> 
>>>>> We have ongoing work (hopefully to be ready in 1-2 weeks) for increasing the cseq number is a sip-wise manner. Just keep an eye on the mailing list.
>>>>> 
>>>>> Regards,
>>>>>  Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>> On 25.08.2014 04:47, Satish Patel wrote:
>>>>>> I am seeing following and all transaction has CSeq: 2 INVITE, I have notice one thing asterisk asking for 407 but opensips never send any challenge response 
>>>>>> 
>>>>>> Opensips ---> INVITE ---> Asterisk
>>>>>> Asterisk -----> 407 ------> Opensips   (Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a710e79".)
>>>>>> Opensips ----> ACK -----> Asterisk
>>>>>> 
>>>>>> Here opensips challenging SIP client  and saying giving try to asterisk and then following
>>>>>> 
>>>>>> Opensips ----> INVITE ---> Asterisk
>>>>>> Opensips ----> INVITE ----> Asterisk 
>>>>>> Opensips ----> INVITE ----> Asterisk 
>>>>>> 
>>>>>> After 3 tries opensips send SIP client 408 Request timeout.. 
>>>>>> 
>>>>>> 
>>>>>> On Sun, Aug 24, 2014 at 4:26 PM, Stefano Pisani <stefano.pisani at omnianet.it> wrote:
>>>>>>> Check if the cseq was incremented by one in the second try.
>>>>>>> Use ngrep.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Il 24/08/2014 22.24, Satish Patel ha scritto:
>>>>>>>> 
>>>>>>>> Hi,
>>>>>>>> 
>>>>>>>> my Opensips (UAC) registered to PSTN gateway and now i am trying to call using my SIPphone which is                                       register to opensip but no success. I am getting 407 Proxy authentication issue..  I am using following method but it didn't work. I need solution badly.. 
>>>>>>>> 
>>>>>>>> PSTN gateway sending 407 Proxy auth and then my Opensip sending 407 proxy auth to SIP phone.  
>>>>>>>> 
>>>>>>>> Does anyone has any working example or some kind of document? I haven;t see any single doc anywhere in Internet about uac_auth  issue
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> modparam("uac","credential","username:domain:password")
>>>>>>>> 
>>>>>>>> route {
>>>>>>>> ....
>>>>>>>> 	    t_on_failure("2");
>>>>>>>> 	    t_relay( "udp:ip_addr:5060" );
>>>>>>>> ...
>>>>>>>> }
>>>>>>>> 
>>>>>>>> failure_route[2] {
>>>>>>>>      uac_auth();
>>>>>>>>      t_relay("udp:ip_addr:5060");
>>>>>>>> }
>>>>>>>>  
>>>>>>>> 
>>>>>>>> 
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>>>>>>> 
>>>>>>> 
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>>>>>> 
>>>>>> 
>>>>>> 
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