[asterisk-users] Exchange UM Play on Phone

[Ryan Wagoner]

On Sat, Jul 24, 2010 at 9:25 PM, Ryan Wagoner <rswagoner at gmail.com> wrote:
> On Sat, Jul 24, 2010 at 12:30 PM, Ryan Wagoner <rswagoner at gmail.com> wrote:
>> On Sat, Jul 24, 2010 at 12:07 PM, Ryan Wagoner <rswagoner at gmail.com> wrote:
>>> I haven't been successful in getting this to work. The issue looks to
>>> be that Asterisk is wanting peer authentication for the invite request
>>> as it sends back 401 Unauthorized.  I am using FreePBX 2.7 and have
>>> tested both Asterisk 1.6.1.18 and 1.6.2.9. My trunk settings are
>>>
>>> type=peer
>>> transport=tcp
>>> qualify=yes
>>> insecure=port,invite
>>> host=10.10.1.31
>>> context=from-internal
>>>
>>> Here is snippets of the SIP debug output. I added in the debug "Peer
>>> has insecure flags" to see what was happening.
>>>
>>> INVITE sip:2001 at voip.mydomain.net;user=phone SIP/2.0
>>> FROM: ""<sip:2001 at exch.testdev.local;user=phone>;epid=079E8F8013;tag=849256682
>>> TO: <sip:2001 at voip.mydomain.net;user=phone>
>>> ...
>>> Sending to 10.10.1.31 : 19219 (no NAT)
>>> Using INVITE request as basis request - 5c97e0f0-5456-4b82-a7f4-e7f2adeba338
>>> Found peer '2001' for '2001' from 10.10.1.31:19219
>>> Peer has insecure flags no
>>>
>>> SIP/2.0 401 Unauthorized
>>>
>>> Due to Exchange making the call from / to the same valid extension
>>> Asterisk is wanting authentication for the 2001. I thought by using
>>> host and insecure in the trunk settings if the from address matched
>>> the host it would use that as the peer. Alternatively I couldn't find
>>> the option to tell Exchange to make the call from a different
>>> extension. In looking at an anonymous call Asterisk doesn't have a
>>> peer for the from number so it looks in from-sip-external.
>>>
>>> INVITE sip:1112223333 at voip.mydomain.net SIP/2.0
>>> From: "1112223333" <sip:1112223333 at voip.remotedomain.com>;tag=as1c8404f3
>>> To: <sip:2223334444 at voip.mydomain.net>
>>> ...
>>> Sending to xxx.xxx.xxx.xxx : 5060 (no NAT)
>>> Using INVITE request as basis request -
>>> 29989544375bf8a162da163d1d9df8bd at voip.remotedomain.com
>>> No matching peer for '1112223333' from 'xxx.xxx.xxx.xxx:5060'
>>> Looking for 2223334444 in from-sip-external (domain voip.mydomain.net)
>>>
>>> Thanks,
>>> Ryan
>>>
>>
>> Looks like I just answered my own question. You can't have a device
>> that matches the user extension. With it configured like this the
>> invite from won't match a SIP peer and it will default to IP lookup.
>>
>> Using INVITE request as basis request - 413feda8-186c-4fe1-a82d-eb074be527e1
>> Found peer 'exchange-vm' for '2001' from 10.10.1.31:63436
>> Peer has insecure flags port,invite
>> Looking for 2001 in from-internal (domain voip.mydomain.net)
>>
>> Ryan
>>
>
> There has got to be a better solution to this involving the invite
> from field peer domain. It looks like find_peer just matches on the
> name and ignores the domain. If domain support is enabled shouldn't we
> only find SIP peers if the from domain on the invite matches one in
> the list? The sip invites I have looked at from Polycom and Linksys
> devices put userid at registrationserver for the from. Or am I missing
> something that this would break?
>
> Ryan
>

I have developed a patch that checks the invite from field domain
against the domain list when domain support is enabled.

https://issues.asterisk.org/view.php?id=17700

Ryan