Re: [asterisk-users] Re: Dynamic Codec Selection
Hi, The PSTN connection is via a zaptel card, rather than a sip peer. With thanks, Tim On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which a re handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? snip I responded: yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. On 2006-10-25 03:31:39 -0700, Wildheart [EMAIL PROTECTED] said: Hi Marty, By the outside world, I mean the PSTN connection. I am still intereste d in how you would set this up. Can you paste in a sample config? One internal phone from SIP.conf: ; ; SIP entry for users test rig [2004] type=friend secret=footest dtmfmode=inband ; my stupid PSTN gateway doesn't like rfc2833 auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw allow=g729 context=autocontext callerid=Alton Wireless Phone 2004 Another internal extension ; IAX entry for user karma [3000] type=friend secret=testfoo auth=md5 host=dynamic disallow=all allow=ulaw allow=g729 context=karma callerid=Karma206500 Ok, now these two extensions when one calls the other should use uLaw. Now here is my extension for my PSTN gateway: ; ; SIP entry for user (FXO) [2003] type=friend secret=testPSTN dtmfmode=inband auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=g729 context=autocontext callerid=Alton Qwest Line2065551183 Depending how your PSTN is setup the last bit could be quite different, but the premise is the same. Since the PSTN only allows g729, this will force other connections to that also. Of course you need to be sure your devices support this, or else you will need to buy licenses for G729 to transcode, which is also a significant hit for CPU. Further more, this only makes sense to do if your PSTN calls are being terminated by someone OFF your local network. If your PSTN calls (like mine) are being routed to a local gateway, then using ulaw should be ok also (it's your network, make it work!). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dynamic Codec Selection
On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which are handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? The idea is to reduce the bandwidth to the server for the majority of calls, but get good quality on internal calls. With thanks, yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dynamic Codec Selection
Hi Marty, By the outside world, I mean the PSTN connection. I am still interested in how you would set this up. Can you paste in a sample config? With thanks, Tim On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which are handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? The idea is to reduce the bandwidth to the server for the majority of calls, but get good quality on internal calls. With thanks, yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Dynamic Codec Selection
On 2006-10-24 06:44:01 -0700, Wildheart [EMAIL PROTECTED] said: Hi, Does anyone know a what to use a different codec for calls which a re handset to handset (eg, G711) then when we have calls to the out side world (via an asterisk server) to use a different codec(eg, G729)? snip I responded: yes, this is simple, just make it so the extensions allow both g729 and ulaw, and set your outside world is g729. On 2006-10-25 03:31:39 -0700, Wildheart [EMAIL PROTECTED] said: Hi Marty, By the outside world, I mean the PSTN connection. I am still intereste d in how you would set this up. Can you paste in a sample config? One internal phone from SIP.conf: ; ; SIP entry for users test rig [2004] type=friend secret=footest dtmfmode=inband ; my stupid PSTN gateway doesn't like rfc2833 auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw allow=g729 context=autocontext callerid=Alton Wireless Phone 2004 Another internal extension ; IAX entry for user karma [3000] type=friend secret=testfoo auth=md5 host=dynamic disallow=all allow=ulaw allow=g729 context=karma callerid=Karma206500 Ok, now these two extensions when one calls the other should use uLaw. Now here is my extension for my PSTN gateway: ; ; SIP entry for user (FXO) [2003] type=friend secret=testPSTN dtmfmode=inband auth=md5 host=dynamic nat=yes canreinvite=no disallow=all allow=g729 context=autocontext callerid=Alton Qwest Line2065551183 Depending how your PSTN is setup the last bit could be quite different, but the premise is the same. Since the PSTN only allows g729, this will force other connections to that also. Of course you need to be sure your devices support this, or else you will need to buy licenses for G729 to transcode, which is also a significant hit for CPU. Further more, this only makes sense to do if your PSTN calls are being terminated by someone OFF your local network. If your PSTN calls (like mine) are being routed to a local gateway, then using ulaw should be ok also (it's your network, make it work!). Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
