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

Vlad Paiu vladpaiu at opensips.org
Mon Aug 25 12:34:22 CEST 2014


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 
> <mailto: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 <mailto: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  <mailto:Users at lists.opensips.org>
>>>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
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>>>
>>>
>>>
>>>
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>>
>
>
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