Voice over IP

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Need help, with calling number

Hy guys, I have this problem with one of clients, he is sending a call to my cisco AS5300, the call is getting translated well, and goes out to the other switch where it has to get billed, but on another switch the calling number is comming wrong or ...

need advise about radius,and billing soft

Hi guys, please advise where can I find or what is the best cisco AS5300 management and billing software? I want to connect few callshops to my AS5300 and also do litle carier to carrier busines, wich free or easy to find Radius server or software is...

Cisco H323 MGW and DTMF relay behaviour

Hello Just want to have a technical position related to MGW H323/DTMFThe Cisco H.323 MGW sends its DTMF capacities in the H.245 Terminal Capability Set (TCS) as described in the general configuration (under voice service voip).The result of this is t...

mjoleroch by Beginner
  • 2 replies
  • 0 Helpful votes

Personal Directory PreDial

We have migrated from CCM4.1.2 to 5.1.2. As Personal Directory is now incorporated within CCM5 and is not a service anymore as in CCM4, how do we specify predial digit to be applied to the directory and fast dial entries as they dial out?

boquiason by Beginner
  • 2 replies
  • 0 Helpful votes

HSI rejects incoming calls from H323

Network sheme:ss7network-> as5350-> PGW2200 (v9.4.1)->-> HSI(4.1)-> H.323 networkCalls from SS7 to H323 network are completed successfully. But if i try to make call in back direction, from h323 to PGW2200 i getting "No_route_to_destination" from HS...

VoIP Module

The Cisco High-Density Analog and Digital Extension Module for Voice and Fax (EVM-HD) contains FXS and FXO ports ?Or I need to buy additional modules with these ports ?

SIP-UA behaviour on Cisco ISR router

Is there there a way to get the sip-ua in the ISR to react to, for example 5xx or 4xx responses from returned after and INVITE? i.e. can/will the sip-ua fail-over to an alternate host/target returned in DNS SRV resource record on receipt of a 5xx o...

ozmark by Beginner
  • 5 replies
  • 0 Helpful votes


Hello, I am trying to find an OID that keeps track of SIP calls in a voice gateway. Does anyone know of one? Any suggestions on how to track active calls besides snmp?