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

Satish Patel satish.txt at gmail.com
Mon Aug 25 20:34:47 CEST 2014


Update, look like 407 error gone and now i am seeing following error

opensips[3767]: ERROR:uac:uac_auth: Unable to extract authentication info


I have following value set and those are correct, realm is asterisk default

modparam("uac_auth","credential","username:asterisk:mypassw0rd#$")




On Mon, Aug 25, 2014 at 10:59 AM, Satish Patel <satish.txt at gmail.com> wrote:

> Perfect!!! just resync code from repo and look like it compile
> successfully!!
>
> I am going to give it a shot and update you soon!
>
>
> On Mon, Aug 25, 2014 at 10:36 AM, Vlad Paiu <vladpaiu at opensips.org> wrote:
>
>>  Hello,
>>
>> Sorry about that - failed to commit auth.h , also apply
>>
>>
>> https://github.com/OpenSIPS/opensips/commit/fc2ed8ace7040734f5c1b11fe235478014817de5
>>
>>
>> Best Regards,
>>
>> Vlad Paiu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>> On 25.08.2014 15:26, Satish Patel wrote:
>>
>> 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 Developerhttp://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 Developerhttp://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 Developerhttp://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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