Hello,
I purchased two polycom 330 to test before we deploy FreePBX. Everything works great except one thing.
When you place a call or receive one, you need to switch to the other line and back before you can hear the call.
I am running sip version 2.2.2. I have read that 1.6.7 is the best version to use, but my soundpoint 330 says is incompatible.
I thought this might have to do with progressinband so I set that in /etc/asterisk/sip_additional.conf for testing purposes. I didn’t seem to help.
here are some logs of a phone call:
[Apr 4 15:48:43] channel.c:3107 ast_channel_make_compatible: No path to translate from SIP/Vonage2-0843b0c8(256) to SIP/030-0848e3e8(4)
[Apr 4 15:48:43] WARNING[7921]: channel.c:2734 set_format: Unable to find a codec translation path from ulaw to g729
[Apr 4 15:48:43] WARNING[7921]: channel.c:2734 set_format: Unable to find a codec translation path from ulaw to g729
– SIP/Vonage2-0843b0c8 is making progress passing it to SIP/030-0848e3e8
– SIP/Vonage2-0843b0c8 answered SIP/030-0848e3e8
– Started music on hold, class ‘default’, on SIP/Vonage2-0843b0c8
– Stopped music on hold on SIP/Vonage2-0843b0c8
Thanks you.