Integration BE6000K CUCM with the Router to connect to PSTN
mnour.tamer
Member Posts: 5 ■□□□□□□□□□
Hello guys , I have a question for you & I will appreciate your help .
I am working on upgrading the phone system for our company & I am using the BE6000S .
Actually for whaty has been done until now , I was able to creat the phone & users & associate all together & there is no issues making a VOIP Connection using the Internal Directory .
before deeping in the advanced feature , I was trying to make an outbound call using the FXO ports to connect to the PSTN .
I have tried using H323 , SIP Trunk & MGCP . Unfortunately I wasn't able to make any outbound call .
I am not sure how the MGCP works , but after creating the MGCP Gateway on the CUCM , I was able to download the configuration on my router running the command ccm-manager config
regarding the H323 & SIP I have created them as well on the CUCM & I have created pots dial plan .
I have created the Route Pattern reffering to all of this gateways & Trunks one by one however I am always having the same issue .
by running the command show voice call status , I can see that the dailed number is picked by the right dial peer with right FXO port & the default codec of G711ulaw which I am using in Canada .
however the dailed number is not ringing & the phone itself is giving me kind of beep .
Can you help me passing this issue before I go and configure the rest .
I am working on upgrading the phone system for our company & I am using the BE6000S .
Actually for whaty has been done until now , I was able to creat the phone & users & associate all together & there is no issues making a VOIP Connection using the Internal Directory .
before deeping in the advanced feature , I was trying to make an outbound call using the FXO ports to connect to the PSTN .
I have tried using H323 , SIP Trunk & MGCP . Unfortunately I wasn't able to make any outbound call .
I am not sure how the MGCP works , but after creating the MGCP Gateway on the CUCM , I was able to download the configuration on my router running the command ccm-manager config
regarding the H323 & SIP I have created them as well on the CUCM & I have created pots dial plan .
I have created the Route Pattern reffering to all of this gateways & Trunks one by one however I am always having the same issue .
by running the command show voice call status , I can see that the dailed number is picked by the right dial peer with right FXO port & the default codec of G711ulaw which I am using in Canada .
however the dailed number is not ringing & the phone itself is giving me kind of beep .
Can you help me passing this issue before I go and configure the rest .
Comments
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shodown Member Posts: 2,271Post your SIP or H323 config.
More than likely you need to setup (disable voip security) and make some test calls to see if they come in.
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CUCM SRND 9x/10, UCCX SRND 10x, QOS SRND, SIP Trunking Guide, anything contact center related