i need some suggestion…my current configuration is:
1 trunks sip euteliavoip (only for outgoing calls with hidden caller ID)
1 trunks myISDN IN and OUT ( 2 numbers available on my ISDN device)
5 Extension (IP301 )
Everything works fine, but i’ve a problem:
when i’m at office and i made an outgoing call it’s ok, but if someone calls me on my ISDN number at the same time, he got a free tone but i cannot answer coz i’m busy with the first call.
Even if only one extension is busy (VOIP or ISDN) i need that for every incoming call the system replies with a busy tone.
Is there someone who can give me a suggestion about that problem?
You have to set the call limits for the sip phones if you want them to signal busy when they are in use, so in sip.conf, try set call-limit=2 (two concurrent calls per phone, to be able to transfer the call) and busy-limit=1 (when the phone is in use signal busy) for every sip phone.
I have been struggling for months to get busy-limit working. I am using Asterisk 1.4.21 and busy-limit just seems to be ignored. You set it to 1 and calls keep coming through. Set call-limit to 1 though and works as expected but then you cant transfer.
I just checked the sip.conf of one of our customers, it has limitonpeers set to yes in the general section, all sip phones have call-limit set to 1; as soon as one call is received/done phones report busy and people are able to do unattended transfe.
I would try set limitonpeers to yes and see what happens.