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

[Satish Patel]

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");
>>>>>> }
>>>>>>  
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>> 
>>>>> 
>>>>> _______________________________________________
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>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> 
>>>> 
>>>> 
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>> 
>> 
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