Olivier
2007-Jul-07 14:06 UTC
[asterisk-users] Which features are lost when canreinvite is turned on ?
Hi, My setup is : PSTN --------- ISTP Network ----------- Router ------------- Asterisk ---------- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I think I should have those phones directly exchanging RTP data with ITSP media gateway, without passing through Asterisk server, with canreinvite = yes option. Before, trying this, I'm wondering which features I would loose in the process ? Will I keep the ability to : - record CDR, - listen to DTMF tones - ... What do you think ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070707/dee3ac91/attachment.htm
Gordon Henderson
2007-Jul-07 14:57 UTC
[asterisk-users] Which features are lost when canreinvite is turned on ?
On Sat, 7 Jul 2007, Olivier wrote:> Hi, > > My setup is : > PSTN --------- ISTP Network ----------- Router ------------- Asterisk > ---------- SIP Phones > > Phones are located in the same location. > I'm thinking about installing new phones in other locations (small agency, > home workers), registering those phones to the same Asterisk server. > > As every location has DSL access, I think I should have those phones > directly exchanging RTP data with ITSP media gateway, without passing > through Asterisk server, with canreinvite = yes option. > > Before, trying this, I'm wondering which features I would loose in the > process ?The ability to pass audio between the endpoints if they are behind NAT firewalls... You might be able to get it work, but I wouldn't bet on it. See: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite http://www.voip-info.org/wiki/index.php?page=Asterisk+Letting+SIP+clients+connect+directly http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions> Will I keep the ability to : > - record CDR, > - listen to DTMF tones > - ... > > What do you think ?I think it's challenging when NAT is involved! Gordon
Anthony Francis
2007-Jul-09 14:01 UTC
[asterisk-users] Which features are lost when canreinvite is turned on ?
Olivier wrote:> Hi, > > My setup is : > PSTN --------- ISTP Network ----------- Router ------------- Asterisk > ---------- SIP Phones > > Phones are located in the same location. > I'm thinking about installing new phones in other locations (small > agency, home workers), registering those phones to the same Asterisk > server. > > As every location has DSL access, I think I should have those phones > directly exchanging RTP data with ITSP media gateway, without passing > through Asterisk server, with canreinvite = yes option. > > Before, trying this, I'm wondering which features I would loose in the > process ? > Will I keep the ability to : > - record CDR, > - listen to DTMF tones > - ... > > What do you think ? > > Regards > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersYou would never lose CDR's because of this feature, and your DTMF should be out of band (in sip messages) anyway. A re invite really just makes the audio connect directly between the sip endpoints in a connection, the sip proxies still receive messages. To understand this better you should read this document: http://www.ietf.org/rfc/rfc2543.txt Hope this helps, Anthony