Ringback issues on MGCP

I have an MGCP gateway configured it is a UTStarcom IAN-02-EX . I configured this solely to learn as it does do SIP.

Making a call to another extension, no ringback is heard although the call rings the other extension and can talk normally

I heard there were some changes in default behavior with callprogressinband and the default setting between asterisk 11 and asteris 13 but can not find reference to it.

Oddly enough on my SIP devices when the Cisco7960SIP calls The Grandstream Handytone 503 FXS port it gets no ringback either. The Cisco 7960 does Ringback just fine to Uniden UIP200. I mention these SIP devices as it may be related to the MGCP ringback issue.

Asterisk 13.14.1~dfsg-2+deb9u4 running on Raspberry Pi 3

MGCP has been community supported for quite some time and I haven’t heard of anyone in years using it. To that end you may be on your own in this regard.

I always wonder why people post replies like this. It is not helpful in any way, and you are more or less saying you know nothing about it. I suggest you therefore not reply.

I am not on my own while this forum exists!

I posted it to set some expectations for you, so that you understand that it’s not a commonly if ever used functionality of Asterisk and so that if you don’t receive replies you’ll know why. I’ll not post any further.

1 Like

I will post here the work-around for ringback problems I came up with if anyone else should need it.

To clarify I was not getting ringback when calling from any MGCP device (extension) to any SIP or MGCP device (extension) . SIP to SIP and SIP to MGCP ws not affected.

I created a 2 minute long ringback audio file, and a 30 second busy (engaged) tone audio file.

I created custom music on hold for ringback and another for busy (engaged)

musiconhold.conf

[ringback]
mode=files
directory=/var/lib/asterisk/ringback

[engaged]
mode=files
directory=/var/lib/asterisk/engaged

Since my MGCP devices are extensions 5921 and 5922 and all SIP devices are 590X and 591X I came up with the following

extensions.conf (from-internal in my case)

;fake ringback and busy 

; dummy extension rings for 30 seconds while we play busy (engaged) tone
exten => 8888,1,Wait(30)
exten => 8888,2,Hangup()

;MGCP to MGCP dialing
exten => _592X/_592X,1,Dial(MGCP/aaln/${EXTEN:3}@192.168.1.49,,m(ringback))
exten => _592X/_592X,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _592X/_592X,3,Hangup()

;MGCP to SIP dialing
exten => _59[01]X/_592X,1,Dial(SIP/${EXTEN},,m(ringback))
exten => _59[01]X/_592X,2,Dial(Local/8888@from-internal,,m(engaged))
exten => _59[01]X/_592X,3,Hangup()

;Real ringback 
;SIP to MGCP
exten => _592X/_59[01]X,1,Dial(MGCP/aaln/${EXTEN:3}@192.168.1.49)
exten => _592X/_59[01]X,2,Hangup()

;SIP to SIP
exten => _59[01]X/_59[01]X,1,Dial(SIP/${EXTEN},,Tr)
exten => _59[01]X/_59[01]X,2,Hangup()

Of course you can modify the above to accomodate voice mail.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.