[asterisk-users] Registering Multiple SIP Accounts on One Server to Another Server
Eric "ManxPower" Wieling
eric at fnords.org
Wed Oct 10 10:08:08 CDT 2007
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.
More information about the asterisk-users
mailing list