SIP IP Phone Incoming issues

Asterisk Asterisk Support
1

Hi

I have a peculiar issue with Grandstream BT-100 phones. They register alright with my Asterisk server. They can receive Incoming calls and dial Outgoing calls too.

But after a period of time, they just get hung up. They cannot receive any incoming calls, but they can dial outgoing calls. Here is the output of the extension which is dead.

  • Name : 2604
    Secret :
    MD5Secret :
    Context : from-priv
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    CallingPres : Presentation Allowed, Not Screened
    Callgroup : 22
    Pickupgroup : 22
    Mailbox : 2604@device
    VM Extension : asterisk
    LastMsgsSent : 256
    Call limit : 0
    Dynamic : Yes
    Callerid : “2604” <2604>
    Expire : -1
    Insecure : no
    Nat : No
    ACL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Trust RPID : No
    Send RPID : No
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : (Unspecified) Port 0
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 2604
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw,alaw)
    Status : Unmonitored
    Useragent : Grandstream BT110 1.0.8.12
    Reg. Contact : sip:2604@10.120.1.137:64594

I suspect it is something to do with Expire field.

Can anybody had similar sort of issues? Your help in this regard is highly appreciated. Thank you very much in advance.

Also if somebody can redirect me to URL explaining what is the function of each field, I will be very thankful to them…

2

Expire as i recall has to do with registration expiery, that is once the SIP registration is made, how long before it either expires or has to be re-registered.

I am assuming that * has a public IP (or forwarded ports) and the BT1xx phones are behind NATs. In this case, I think qualify=yes may solve your problem. That makes * send a periodic sip message to the phone to keep the NAT mapping alive.

If the phones are on the same network as *, go into the phone setup adn turn off STUN.

If that doesnt help post more info about your network :smile:

3

Thanks for your reply Iron Helix. The * and the phones are on the same network.

Ok I will try disabling STUN. I will post the updated results.

4

STUN is disabled on the phones…still the same result. Extension die after a period of time and do not receive any incoming calls. Please anybody any ideas where to look for the issue?

Thanks in advance…

5

when a phone dies, if qualify=yes for it, do sip show peers and paste the output.

both in * and on the phone, check the registration expiery to make sure they are not too long or too different. Make the max expiery 300 (5 minutes) and default 60 (one minute).

Also what firmware rev are the phones running? Maybe they need an update?