Transfer to voice mail not working on polycom phones

using polycom 550’s, with endpoint manager. transfer to an extension works. can press transfer, number, transfer and everything is good. however cannot do a transfer to voice mail. when pressing transfer and then “*” the screen shows “*sip@server ip address” and then starts appending the extension digits to the last number of the ip address.

using polycom ucs 3.3.4

Looks like a phone thing. Instead of passing the “*” to Asterisk, the phone is intercepting it and performing some function. Take a good look at the phone’s admin manual and you should find your answer.

BF

Try ##*extension

however the issue i am wrestling with is that there is a transfer button on the phone - mostly a training thing but when someone sees a transfer button they tend to push it. i will stare at the templates that the endpoint manager uses to see if i can spot where this is going wrong.

What are your dialing digit maps on the phone dialing plans?

Is *XXX allowed in the dial plan?

It sounds a bit like this - http://www.freepbx.org/forum/freepbx/general-help/unable-to-engage-feature-codes-with

i agree it is a phone issue - i have played with the dial plan in the phone to see if i could find a way to make it work. so far no luck

What is your dialplan?

Mine appears to work… I have polycom 450 & 550, freepbx distro (the 2.9 latest, not 2.10).

i have installed the 3.3.4 software with the latest boot room. and are you using endpoint manager?

I am using endpoint manager with whatever software/bootrom came with endpoint manager. Looks like 3.3.1 sip.ld when I browse through the phone menus.

install the sip.ld into the tftpboot directory?
it did not for this installation and i simply copied the appropriate sip.ld file into the directory. i wonder if there was in issue with the installation?

yes - it did install all the files. Did you install the polycom package and the appropriate firmware package?

i am almost certain this is something on the phone. somewhere there must be a setting that the * triggers. the reason ## works is that it is sent to the pbx. the pbx responds “transfer”. the * causes a url (*@hostip) to be displayed on the phone.
i have added elements to the dial plan on the phone to process the * but they have no effect. looks like more reading in the polycom manual. i have another pbx, same software, same phones and all works perfectly. i have compared the sip.cfg files and the generated config files between the two systems and there is no difference.

the one difference between the two systems is the one having the issue has separate devices and users and the one that works has them combined. i split them so that we could have one extension that rings two phones (phones are in different locations).

weird.

I am using device/user mode - not that it helps the picture…

whatever it is, it happened to all 14 phones connected to this system only.

i should have checked this before, but if i take the phone off hook and dial a * i get the same problem - the screen shows *@pbx-ip and positions the cursor at the end of the string. if the phone is on hook, i can dial a feature code, then pick up the handset and it works fine.

That is the same problem I had before in the post referenced above. Suggest you triple check your dial plan digits (post them)? And make sure the phones are loading them? I sometimes look at /var/log/messages for what files tftp is serving.

[root@pbx tftpboot]# pwd /tftpboot [root@pbx tftpboot]# grep -i 'digitmap=' 000*_reg.cfg dialplan.1.digitmap="[2-9]11|0T|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|[2-9]xxxxxx|*xxx|*xx.T" dialplan.2.digitmap="[2-9]11|0T|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|[2-9]xxxxxx|*xxx|*xx.T" dialplan.3.digitmap="[2-9]11|0T|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx|[2-9]xxxxxx|*xxx|*xx.T"

once i manually fixed the dial plan in the config file things now work. so i still have two issues. first is why the endpoint manager built the wrong dial plan ( dialplan.1.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[1-9]xxT|[2-9]xxxT") and second, i was under the impression that editing the dial plan on the phone itself took precedence over anything config loaded from the host - when i manually entered the correct dial plan on the phone directly via the web interface, the phone would reboot, i could see the updated dial plan but it seemed to be using the dial plan from the host config file.

Not sure where the endpoint manager dialplan came from.

Web changes to the phone are overwritten by the info downloaded from the server from what I can tell. From my experience, the only changes that are sticky are the ones made from the phone interface, which are uploaded by the phone into the overrides directory. That is stuff like brightness, etc. I quit using the web interface a long time back, once I figured out the tftp magic.