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Product page
Configuration guides
Call direction and tranlation translation section
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voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip |
Alow Allow or ban directions you want to use.
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Here can be set protocol parameters like slow start and fast start for H.323. Call start interwork allows translation between these modes. Protocol setting can changes be changed for particular dial peer bey i.e. by means of voice-class h323 commands or by appropriate commands directly in dial peers
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voice class h323 500 call start fast |
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dial-peer voice 211 voip
destination-pattern 91........
voice-class codec 200
session protocol sipv2
session target ipv4:5.6.7.8
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H.323
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dial
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peer
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that
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uses
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RAS
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(configured
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gatekeeper).
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dial-peer voice 1 voip |
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destination-pattern 10......... |
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voice-class codec 200 |
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session target ras |
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H.323
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dial
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peer
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that
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routes
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to
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another
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gw
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or
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endpoint
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dial-peer voice 1 voip |
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destination-pattern 11......... |
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voice-class codec 200 |
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voice-class h323 500 |
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session target ipv4:9.8.7.6 |
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Specific
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h323
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class
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is
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configured
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to
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change
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some
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parameters.
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SIP
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UA
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settings
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* sip-ua |
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nat symmetric role passive |
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nat symmetric check-media-src |
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sip-server dns:domain.dom |
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Here
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is
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defined
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sip
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server
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(used
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in
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dial
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peers)
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and
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various
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sip
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parameters.
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Gatekeeper settings
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interterface Loopback0 |
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description MGMT Loopback |
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ip address 1.2.3.4 255.255.255.255 |
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h323-gateway voip interface |
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h323-gateway voip id GK1 ipaddr 1.1.1.1 1718 priority 100 |
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h323-gateway voip id GK2 ipaddr 1.1.1.2 1718 |
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h323-gateway voip h323-id IP2IPGW |
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h323-gateway voip tech-prefix 1# |
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h323-gateway voip bind srcaddr 1.2.3.4 |
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Gatekeeper setting that settings for ras session target in dial peers with pramary primary and secondary gatekeeper. It is not necessary to configure it on Loopback interface.
There seems to be a BUG that affects H.323 slowstart to SIP calls if sip server is not set by ipv4:1.2.3.4.
OS specific help
N/A - There is only CISCO IOS
Validation, confirmation tests
Route call through IP2IP gw.
If you don't use sip server and gatekeeper, set dial peer destinations (session-target) directly to endpoints and call each other ( i.e. sip:numberofh323endpoint@ipofip2ip)