I recently inherited the duties of managing an asterisk phone system. I can call in and receive our automated teller. When I transfer to an extension, I can user the end user but they cannot hear me. Additionally, I am unable to call out using any of the phones that we have on the network.

I know it’s incredibly vague, but I have no clue where to even start looking at what the issue is. I modified the configuration files to a new network IP set we have, however, I don’t know how to update the sip.ld file so it loads properly on the Polycom phones we have.

Any assistance is greatly appreciated.


you probably need to modify /etc/asterisk/sip.conf and reset externip= and localnet= with your new network info.

then make sure that udp port 5060 (if you do calls via SIP outside your network) and the RTP range (/etc/asterisk/rtp.conf) are forwarded to your * box…

I think the issue is that since the IP address changed for the server, the phones don’t know where to locate the sip.ld file. Is there a way to deploy that from the server?

Polycoms can be configured in a number of ways, probably via TFTP. Most phones (incl polycom) can get TFTP info via DHCP, so its possible that the answer lies in your DHCP server. Setting it to provide the new IP for TFTP (option 66 as i recall) may solve the first part of the problem.

part 2 will be to get the polycom config files (probably in /tftpboot) to reference the new IP… this you can probably do by text editing them with nano -w or something…

Alright, so now I think I have my head wrapped around this. The former employee who managed this was left right around the time that a firm we shared office space with, and I believe that we were more or less a mini-lan off of their network. I bet they (the other firm) have the TFTP server and were the primary DHCP hosts. That being said, I have no way to deliver the configuration file out to the phones, hence they have old configuration files. Since, again, I’m new to this… what are my options?