[Users] Using uac_replace_from to munge username between ser and Asterisk?

Barry Flanagan barryf-lists at flanagan.ie
Fri Feb 3 23:16:48 CET 2006


Bogdan-Andrei Iancu wrote:
> Hi Barry,
> 
> indeed, messing with the FROM URI may brake things that rely on it...
> not sure about mediaproxy, but nathelper uses only the from-tag, which 
> is not replaced....I would say it should work..have you tried?

Yes, nathelper on its own appears to be ok.

Is nathelper on its own generally sufficient for a public SIP server 
where most clients are behind NAT?

Thanks.

-Barry

> 
> regards,
> bogdan
> 
> Barry Flanagan wrote:
> 
>> Barry Flanagan wrote:
>>
>>> Bogdan-Andrei Iancu wrote:
>>>
>>>> if you enable auto from_restore_mode, you do not need to perform any 
>>>> restore from script. Just replace the from in the initial INVITE and 
>>>> this is it - all replies and sequential request would be auto fixed 
>>>> (restore/replace).
>>>
>>>
>>>
>>>
>>> Aha! Thank you very much  - that seems to have done the trick!
>>
>>
>>
>> Hmm, OK, it did work until I started using nathelper and mediaproxy. 
>> With mediaprixy I get no audio, and Asterisk is retransmitting
>>
>> It appears that mediaproxy is looking for the "unmunged" username. 
>> Below is the mediaproxy log for this call. It is expecting 
>> from:sipps2 at sip.domain.com, whereas asterisk is sending 
>> sipps2_domain.com at sip.domain.com
>>
>> Feb  3 11:25:05 www1 mediaproxy[7461]: command request 
>> 1647296070-45779966 at XXX.XXX.96.225 
>> XXX.XXX.96.225:12047:audio,XXX.XXX.96.225:12049:video XXX.XXX.96.225 
>> sip.domain.com local sip.domain.com local 
>> Nero=20SIPPS=20IP=20Phone=20Version=202.1.3.25 
>> info=from:sipps2 at sip.domain.com,to:0863854334 at sip.domain.com,fromtag:622fb836,totag: 
>>
>> Feb  3 11:25:05 www1 mediaproxy[7461]: session 
>> 1647296070-45779966 at XXX.XXX.96.225: started. listening on 
>> XXX.XXX.1.16:35194,35196
>>
>>
>> Here Asterisk is retransmitting:
>> Retransmitting #1 (NAT) to XXX.XXX.1.16:5060:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 
>> XXX.XXX.1.16;branch=z9hG4bK9e52.4f879791.0;received=XXX.XXX.1.16
>> Via: SIP/2.0/UDP 
>> XXX.XXX.96.225:12046;branch=z9hG4bKnp1643392953-45a6e6feXXX.XXX.96.225;rport=12020 
>>
>> Record-Route: <sip:XXX.XXX.1.16:5060;nat=yes;ftag=61f4a3ce;lr=on>
>> From: ""Barry Flanagan"" 
>> <sip:sipps2_domain.com at sip.domain.com>;tag=61f4a3ce
>> To: <sip:0863854334 at sip.domain.com>;tag=as53b2855b
>> Call-ID: 1643422664-49d79852 at XXX.XXX.96.225
>> CSeq: 2 INVITE
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Contact: <sip:0863854334 at XXX.XXX.1.68>
>> Content-Type: application/sdp
>> Content-Length: 201
>>
>>
>>  Any idea?
>>
>> Thanks.
>>
>> -Barry Flanagan
>>
>>
>>
>>>
>>> Regards,
>>>
>>> -Barry Flanagan
>>>
>>>>
>>>> regards,
>>>> bogdan
>>>>
>>>> Barry Flanagan wrote:
>>>>
>>>>> Bogdan-Andrei Iancu wrote:
>>>>>
>>>>>> Hi Barry,
>>>>>>
>>>>>> have you set auto from restoring? See:
>>>>>>    http://openser.org/docs/modules/1.1.x/uac.html#AEN75
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Yes, but I am not sure where it is supposed to go.
>>>>>
>>>>> I have the following in just before relaying to Asterisk:
>>>>>
>>>>>     rewritehostport("XXX.XXX.XXX.XXX:5060");
>>>>>     uac_replace_from("$fn","sip:$au_$ar@$fd");
>>>>>     append_hf("P-hint: GATEWAY\r\n");
>>>>>     t_relay("udp:XXX.XXX.XXX.XXX:5060");
>>>>>
>>>>>
>>>>> and I put in uac_restore_from(); just after the record_route()
>>>>>
>>>>>
>>>>> with all the other modparams I have:
>>>>>
>>>>> modparam("uac","from_restore_mode","auto")
>>>>>
>>>>>
>>>>> Thanks for the help.
>>>>>
>>>>> -Barry
>>>>>
>>>>>
>>>>>> regards,
>>>>>> bogdan
>>>>>>
>>>>>> Barry Flanagan wrote:
>>>>>>
>>>>>>>
>>>>>>> So, the only way around it that I can see is to somehow have 
>>>>>>> OpenSER change the username to username_domain so that each will 
>>>>>>> be unique.
>>>>>>>
>>>>>>> It looks like uac_from_replace should handle this. I have tried 
>>>>>>> it, and I can see that Asterisk does in fact get 
>>>>>>> user_domain at domain in the first invite, but thereafter for some 
>>>>>>> reason OpenSER changes it to just _ at domain for subsequent requests.
>>>>>>>
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> -Barry
>>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>>
>>
>>
> 


-- 

-Barry Flanagan




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