darwin.solano at gmail.com
2009-Mar-24 22:08 UTC
[asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
Test ------Mensaje original------ De: tracinet Remitente:asterisk-users-bounces at lists.digium.com Para:Asterisk Users Mailing List - Non-Commercial Discussion Responder a:Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP Enviado: 6 Mar, 2009 5:55 PM Basically, Server 1 is the main customer PBX where we have multiple customers running (each on their own virtual PBX separated by their contexts).? Each customer has their own accountcode that we use to track calls for billing purposes, etc.? The customer uses a SIP phone to register to Server 1 and sends calls to it.? Server 1 in turn, passes the calls to Server 2 which is connected to various SIP providers and T-1's, etc. for termination to the PSTN.?? In the following sip configuration, calls work perfectly, except that the caller ID gets passed as the value from "fromuser" instead of the numeric value we set via the Set(CALLERID(num)=5555555555) command.? In other words, the fromuser overrides the caller ID value.? If we remove the "fromuser" in the sip configuration, calls work great and caller ID is passed, BUT all calls land in the customerb context on Server 2 since that is the last SIP entry in sip.conf that has a host entry set to "192.168.0.11" which is the IP of Server 1. Server 1 (192.168.0.11) sip.conf [general] disallow = all allow = ulaw port = 5060 context = incoming maxexpirey=3600 defaultexpirey=300 canreinvite=no dtmfmode=auto nat=yes ; Customer A Outbound SIP [customera-out] context=customera type=friend username=customera-out fromuser=customera-out secret=aaaa host=192.168.0.12 canreinvite=no accountcode=customera amaflags=billing dtmfmode=auto ; Customer A SIP Phone Account [customera101] context=customera type=friend username=customera101 secret=1234 host=dynamic canreinvite=no mailbox=101 at customera nat=yes qualify=yes callerid="John Smith" <101> accountcode=customera amaflags=billing dtmfmode=rfc2833 ; Customer B Outbound SIP [customerb-out] context=customerb type=friend username=customerb-out fromuser=customerb-out secret=bbbb host=192.168.0.12 canreinvite=no accountcode=customerb amaflags=billing dtmfmode=auto ; Customer B SIP Phone Account [customerb101] context=customerb type=friend username=customerb101 secret=1234 host=dynamic canreinvite=no mailbox=101 at customerb nat=yes qualify=yes Enviado desde mi m?vil BlackBerry Orange.
Matt Riddell
2009-Mar-24 22:20 UTC
[asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On 25/03/2009 11:08 a.m., darwin.solano at gmail.com wrote:> Testfailed :) -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)