On Mon, Dec 18, 2017 at 9:04 AM, Dan Cropp <[email protected]> wrote: > Thanks George > > > > I originally didn’t have the 1002@ for the identify. Changed that when > things were not working. I changed it back. > > > > Unfortunately, the system I am connecting with doesn’t seem to support the > line support. Looking at the SIP packets, I see Asterisk send it. > Unfortunately, they do not send the line information as part of the > INVITE. I checked with some developers of that system and they do not know > anything about the line setting. > > Is there any rfcs I could refer them to? >
Yeah, I've found that some providers do and some providers don't. https://tools.ietf.org/html/rfc3261#section-19.1 An implementation MUST include any provided transport, maddr, ttl, or user parameter in the Request-URI of the formed request. If the URI contains a method parameter, its value MUST be used as the method of the request. The method parameter MUST NOT be placed in the Request-URI. *Unknown URI parameters MUST be placed in the message'sRequest-URI*. The identify object also has the capability to match against a specific header and value but it looks like it only tries to match on header if it can't find a match by ip address. Here's some info on it anyway. If you're provider puts something unique and constant in the headers, like a User-Agent string that doesn't change, you can also try using the "match_header" parameter to an identify object. [my_provider] type = identify match_header = User-Agent: Something Unique 1.0.0 endpoint = provider It has to be an exact match though, no wildcards or regular expressions. I opened an issue[1] on separating ip matching from header matching so they can be re-ordered. [1] https://issues.asterisk.org/jira/browse/ASTERISK-27491 > > > > *From:* [email protected] [mailto:asterisk-users- > [email protected]] *On Behalf Of *George Joseph > *Sent:* Thursday, December 14, 2017 10:59 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Is it possible to have two endpoints to > the same IP address where one uses IP based authentication and the other > requires asterisk to register to that system? > > > > > > > > On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp <[email protected]> wrote: > > Currently using PJSIP. First, they want me to get this working with the > existing PJSIP configuration, but then setup a second box using chan_sip > performing similar work. > > > > For PJSIP… > > I currently have an endpoint configured to a system using IP based > authentication. It is configured with a match setting in the endpoint > section. > > All channels coming from that IP address go to this endpoint. > > > > They want me to keep this endpoint, but add a new endpoint where we > register with them. > > > > Existing… > > [transport1] > > type = transport > > bind = 0.0.0.0 > > protocol = udp > > > > [1002] > > type = aor > > remove_existing = yes > > contact = sip:[email protected] > > > > [1002] > > type = endpoint > > context = mycontext > > transport = transport1 > > accountcode = 6 > > dtmf_mode = inband > > device_state_busy_at = 48 > > force_rport = no > > identify_by = username > > from_user = 1002 > > disallow = all > > allow = ulaw > > acl = acl1 > > > > [identify112] > > type = identify > > endpoint = 1002 > > match = [email protected] > > > > > > Check this first... identify112 probably failed to load because the match > parameter can only take an ip address > > plus an optional netmask, or a hostname. The '1002@' is invalid. > > > > > > > > > > I setup the registration and the endpoint. > > > > [286] > > type = aor > > remove_existing = yes > > contact = sip:[email protected] > > qualify_frequency = 60 > > > > [auth8] > > type = auth > > username = 286 > > password = yyyyyyyyyyyyyyy > > > > [286] > > type = endpoint > > context = mycontext > > transport = transport1 > > outbound_auth = auth8 > > aors = 286 > > accountcode = 22 > > dtmf_mode = inband > > device_state_busy_at = 48 > > force_rport = no > > disallow = all > > allow = ulaw > > acl = acl1 > > > > [registration3] > > type = registration > > transport = transport1 > > client_uri = sip:[email protected] > > server_uri = sip:xxx.xxx.xxx.xxx > > contact_user = 286 > > outbound_auth = auth8 > > expiration = 3600 > > > > The registration for the second endpoint works fine. However, when I call > through the other system for 286, it is failing. For the INVITE from the > other switch, the from_user varies depending on who is calling. Asterisk > logs report “No matching endpoint found” when it processes the INVITE for > 286. > > > > I believe the reason INVITEs work for the other channel is because they > are programmed to support the match for this IP address. > > > > Can anyone offer some suggestions? > > > > You may be able to use the 'line and 'endpoint' registration parameters... > > [registration3] > > type = registration > > ... > > line = yes > > endpoint = 286 > > > > This causes asterisk to put the encoded endpoint name in the outgoing > Contact header. If the provider properly echos back Contact parameters > when sending responses or new requests, asterisk will use the line > parameter to match an endpoint. I'll have to double check but I believe we > do that BEFORE checking any identify object for a match. > > > > > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > George Joseph > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
