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

[Satish Patel]

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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>>>> Users at lists.opensips.org
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>>> 
>>> 
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>> 
>> 
>> 
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