Call Forwarding not Working

Hello Everybody.

I’ve been using my recently installed FreePBX without any problems, but when I was trying to configure call forwarding feature codes I noticed it doesn’t works, on the phone I hear nothing and the Asterisk CLI shows the following:

– Executing [*[email protected]:8] GotoIf(“SIP/5230-000001c4”, “1?startread”) in new stack
– Goto (from-internal,*72,7)
– Executing [*[email protected]:7] Read(“SIP/5230-000001c4”, “toext,ent-target-attendant&then-press-pound,”) in new stack
[2012-12-11 15:43:39] WARNING[20222]: file.c:663 ast_openstream_full: File ent-target-attendant does not exist in any format
[2012-12-11 15:43:39] WARNING[20222]: file.c:954 ast_streamfile: Unable to open ent-target-attendant (format 0x8 (alaw)): No such file or directory
[2012-12-11 15:43:39] WARNING[20222]: file.c:663 ast_openstream_full: File then-press-pound does not exist in any format
[2012-12-11 15:43:39] WARNING[20222]: file.c:954 ast_streamfile: Unable to open then-press-pound (format 0x8 (alaw)): No such file or directory
– User disconnected
– Executing [*[email protected]:8] GotoIf(“SIP/5230-000001c4”, “1?startread”) in new stack
– Goto (from-internal,*72,7)
– Executing [*[email protected]:7] Read(“SIP/5230-000001c4”, “toext,ent-target-attendant&then-press-pound,”) in new stack
[2012-12-11 15:43:39] WARNING[20222]: file.c:663 ast_openstream_full: File ent-target-attendant does not exist in any format
[2012-12-11 15:43:39] WARNING[20222]: file.c:954 ast_streamfile: Unable to open ent-target-attendant (format 0x8 (alaw)): No such file or directory
[2012-12-11 15:43:39] WARNING[20222]: file.c:663 ast_openstream_full: File then-press-pound does not exist in any format
[2012-12-11 15:43:39] WARNING[20222]: file.c:954 ast_streamfile: Unable to open then-press-pound (format 0x8 (alaw)): No such file or directory
– User disconnected

And it loops there until I hang up the call.

It seems to be missing some audio files with the instructions. I’ve tried to input the extension number then #, then the other extension and then # again but nothing happens.

I’ve reinstalled the call forward module several times and also the system recordings module to see if I could get that fixed that way but haven’t been able to.

Can this be caused by the voicemail being active?

I hope you can help me soon with this.

Thanks

I experienced today the same problem described in this post. For those facing the same issue the problem appears to be missing sound files. Just need to recompile asterisk and in the menuselect step select not only the core sound files but also the Music on Hold and Extra Sounds packages.
This action solved the problem for me.