1. dial-peer voice 500 voip I use this configuration for inbound to asterisk.
2. dial-peer voice 510 pots description Fancy PRI - Outgoing huntstop destination-pattern .T direct-inward-dial forward-digits 10 And use this configuration for outbound from asterisk to Cisco 5350 right? On Tue, Nov 25, 2008 at 3:42 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: > My attention to my dial peer. It has nothing about H.323 and much about > SIP. > > A T I F wrote: > > > Alex, > > > > 1 more thing my gateway is configured with H.323 so tell me how can I > > configure it with SIP? > > > > On Tue, Nov 25, 2008 at 3:10 PM, Alex Balashov > > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: > > > > Set up a SIP dial peer and an outbound POTS dial peer. Bear in mind > > that the gateway shunts calls POTS->VOIP and VOIP->POTS by default, > so > > you can use the same destination pattern matching for both in this > > simple scenario, but if it gets any more complicated than that, some > > degree of translation is almost certainly required. > > > > The process can be fairly complex, but the general idea, if you have > > your TDM side set up, is: > > > > dial-peer voice 500 voip > > description Asterisk > > destination-pattern .T > > progress_ind setup enable 3 > > voice-class codec 1 > > session protocol sipv2 > > session target ipv4:ip.addr.of.asterisk > > session transport udp > > dtmf-relay rtp-nte > > no vad > > > > dial-peer voice 510 pots > > description Fancy PRI - Outgoing > > huntstop > > destination-pattern .T > > direct-inward-dial > > forward-digits 10 > > > > > > A T I F wrote: > > > > > Hello, everybody! > > > > > > I need help connecting my Cisco AS5350 to Asterisk. > > > > > > What i want to do is forward all outgoing calls from Asterisk > > server to > > > Cisco AS5350, and from Cisco 5350 to my Asterisk server, using > SIP. > > > > > > How could this be done? > > > > > > Thanks in advance > > > > > > Atif Shahzad > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > -- 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-users > > > > > > -- > > Alex Balashov > > Evariste Systems > > Web : http://www.evaristesys.com/ > > Tel : (+1) (678) 954-0670 > > Direct : (+1) (678) 954-0671 > > Mobile : (+1) (706) 338-8599 > > > > _______________________________________________ > > -- 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-users > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- 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-users > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > _______________________________________________ > -- 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-users >
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