No IVR listen at device end……SIP phone is working fine

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Asterisk Users 1 Comment

Hi Virendra,

It may be problem for rtp packet port forwarding if u can dial through DID
number.

You need to open rtp port range in firewal. e.g. 10,000 to 20,000 port.

please, write how can you dial call mobile or other devices. e.g. DID
number, PRI number etc.

One thought on - No IVR listen at device end……SIP phone is working fine

  • Hi Rajnikant,

    Foe making outdial we are using VoIP trunk which is working fine. But for
    taking incoming calls in routing to any defined extension we are using DID.

    So for DID incoming we dial DID from out Cisco 79XX then call come to server
    after routed by DID provider to our server. Then our server get the calls
    and then dial SIP extension from here. But we didn’t get any voice or IVR
    option’s just like Press 1 for this…2 for that….
    And extension is also not ring after all…… I am using Elastix for
    routing calls …..But I also test with asterisk dialplan too….

    CLI Output s blow:-