[asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server

[Steve Totaro]

Eric "ManxPower" Wieling wrote:
> Steve Totaro wrote:
>> Eric "ManxPower" Wieling wrote:
>>> Steve Totaro wrote:
>>>> Steve Totaro wrote:
>>>>> I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it 
>>>>> worked fine except for audio issues that I believe are directly related 
>>>>> to IAX2 in version 1.2.x.  I have four PRIs and want a separate context 
>>>>> for each going into the PBX.  This worked very well with IAX.
>>>>>
>>>>> I want to use SIP to see if the audio issues are eliminated but Asterisk 
>>>>> does not seem to like multiple SIP account from one box to another (four 
>>>>> to be exact)
>>>>>
>>>>> I found this http://www.voip-forum.com/news.php?p=187 which makes me 
>>>>> think this is a known problem.  Unfortunately, the link goes to an error 
>>>>> page.
>>>>>
>>>>> I have tried ever combination of credentials and setting in SIP conf but 
>>>>> the calls still fail.  I tried friend, user, insecure=very, username, 
>>>>> from user, and anything else I could think of.
>>>>>
>>>>> Is there something I am missing or a workaround for this issue?
>>>>>
>>>>> PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP--> PBX 
>>>>> (calls fail)
>>>>> PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf--> PBX (calls 
>>>>> work)
>>>>>
>>>>> Thanks,
>>>>> Steve Totaro
>>>> I think I may have figured out my own issue.  Since I am creating 
>>>> multiple SIP peers on two boxes that point to each other, I need to 
>>>> define separate ports for each one.  Anyone know if that is the case? 
>>>> Makes sense to me but I cannot try it on the live server and my dev 
>>>> boxes are all doing other things.
>>> no.  It might be the case if you had multiple SIP clients behind the 
>>> same NAT router connection to a non-local Asterisk box.
>>>
>>> The userid and password that is sent with the call should make it hit 
>>> the correct sip.conf entry.  Perhaps you are doing something silly in 
>>> your sip.conf configs.
>>>
>> Perhaps I am, let's hope so.  This was my latest attempt to get it to 
>> work.  The other server looks identical except the host IP.
>>
>> [general]
>> ;bindport=5060
>> bindaddr=0.0.0.0
>>
>> [default]
>>
>> [span1]
>> type=friend
>> host=192.168.6.2
>> username=span1
>> secret=xxxx
>> context=to-span1
>> auth=rsa
>> inkeys=span1-2-fast1
>> outkey=fast1-2-span1
>> qualify=yes
>> disallow=all
>> allow=ulaw
>> allow=slin
>> allow=alaw
>> insecure=very
> 
> I don't use RSA auth so I can't comment on that.  My understanding of 
> insecure=very is vague, but you do NOT need it for Asterisk<->Asterisk 
> SIP connections and I suspect that is what is causing your problem.  I 
> recommend against using qualify.
> 

Thanks, I will give those recommendations a try.  If not, I am going to 
re-do their entire setup in a dev environment and then just move it over 
after testing.

Thanks,
Steve totaro