Call Forwarding rings but no audio. Follow me rings but no audio

General Help
1

When call forwarding is set up the phone rings on the other end but when the call is picked up there is no audio. Also Follow me only works if user confirmation is enabled, otherwise it’s the same behavior as Call Forwarding. Does anyone have any ideas?

2

What version of asterisk are you using? I know there was a bug where the call had to be sent to an IVR first, I use a 0 second timeout then go to the extension.

3

I have seen posts on several forums that mention the IVR work-around. I have looked at my extensions.conf:

[macro-stdexten]
exten = s,1,Set(_DYNAMIC_FEATURES=${FEATURES})
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,GotoIf($["${FOLLOWME${ARG1}}" = “1”]?6:4)
exten = s,4,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
…

I am not trying to use FOLLOWME. When call forwarding is set on any handset, the Asterisk system appears to retrieve the new number from the handset and dial it without returning to the dialplan. Do I need to put the IVR before the s,4,Dial line in my dialplan?

Anyone have an example to share?

Thanks

4

Finally found an example. It only took three days–and of course I found it shortly after posting my question.

For any who find this thread, here is my revised extensions.conf:

[macro-stdexten]
exten = s,1,Set(_DYNAMIC_FEATURES=${FEATURES})
exten = s,2,Set(ORIG_ARG1=${ARG1})
exten = s,3,GotoIf($["${FOLLOWME${ARG1}}" = “1”]?8:4)
exten = s,4,Answer
exten = s,5,Playback(silence/1)
exten = s,6,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
…

Regards

5

As I recall it’s not a bug but a result of connecting Asterisk through NAT.
Fix is very simple - forward RTP Port range on the NAT.

6

Also for anyone reading the thread, do not modify extensions-additional.conf

Only filenames containing the work ‘custom’ in /etc/asterisk are not overwritten when you “apply config” in FreePBX.

7

Adding the RTP port range on the firewall did not work in my case. I tried every “solution” I could find until the IVR work-around was the only one left. It was only left until last though because it was difficult finding an example, so I offered one here.

Having said that, the IVR work-around should be everyone’s last resort. It is the least efficient use of resources and every call is “answered” whether the extension called is forwarded or not.

8

You need to open an audio path in the PBX first. The easiest way to accomplish that is to have the inbound route go to an Announcement module that plays no announcement and then forwards on to wherever you normally would send the call. That’ll force Asterisk to answer the call and open an audio path before the call gets forwarded. Asterisk will then generate the ring sound, which your caller should hear before the call gets forwarded. If the caller hears the ringing, but cannot hear the answering party, then the audio loss is on the path from the PBX to the answering party.

9

I am having this issue but can hear two rings first and then it goes blank. Thus the issue seems to be on the path from the PBX to the answering party. Can someone suggest the steps to trouble shoot this problem? Thanks!