Gigaset + Transfer gives a busy message

Hi everybody, let me explain you my situation :

  • I have installed a freepbx distribution (FreePBX Framework,
  • I have defined many extensions, IVR, Trunks, etc … everything is working fine
  • I have bought a Gigaset phone C610IP with 2 phones and one basis plugged on my network. I can receive calls from each phone, which are connected to the same asterisk and with 2 differents extensions

So far everything is great !

The strange think is when I want to do a call forwarding from a phone to another phone. I receive a call from an external number (inbound route) or internal extension (an android phone connected with csipsimple) to a gigaset device. I answer, and I can talk. But when I want to transfer it to the other gigaset device I get a busy message.

I can transfer it to any softphone connected, but not to the second device. Note that my extensions are connected with 2 differents ports (IP are same) :

[2012-08-22 12:41:46] VERBOSE[13900] chan_sip.c: – Registered SIP ‘3304’ at 88.x8x.11.xx:58673
[2012-08-22 12:41:54] VERBOSE[13900] chan_sip.c: – Registered SIP ‘3300’ at 88.x8x.11.xx:46853

When the first device is on call, I can call the second device from any other account (connected with a softphone) or I can call from the second device to anything.

To try to most extreme case I have done the following :

  • gigaset device 1 receives an external call
  • it transfers it to a connected extension via a softphone
  • this softphone transfers it to the gigaset device 2 => Working

When :

  • gigaset device 1 receives an external call
  • it transfers it to the gigaset device 2 => NOT Working

I get the following message :

[2012-08-22 13:33:27] VERBOSE[13900] chan_sip.c: – Got SIP response 486 “Busy Here” back from 88.x8x.11.xx:46853
[2012-08-22 13:33:27] VERBOSE[21512] app_dial.c: – SIP/3300-0000006b is busy
[2012-08-22 13:33:27] VERBOSE[21512] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)

Does somebody has an idea ?

Thanks by advance

Turn off reinvite behavior on PBX and Phone.

Is phone in same network as PBX?

Thanks for your help,

reinvite are deactivated on each extension + globally. From the gigaset device I don’t believe I have such option.

No the asterisk is on another network, on a public one. My 2 phones are on a single basis with one IP. I had to play with ports to make them working.

My softphones are on the same network than the phones.


Yes, reinvite behavior is programmable in FreePBX in both trunk and extension level configuration.

Since the phone is offsite this could be one cause of the issue.

The other possibility since the phone is remote is some sort of network issue. THe fact in only occurs on transfers leads me back to reinvite.

Thanks, but I would have had the same issues with my softphone then. I believe that it’s coming from my gigaset that doesn’t let me to receive 2 calls in the same time from the same source. I believed I could find a workaround with asterisk …

But I don’t have any other idea.

I Suspect your router is overmangling the port translation of two connection on the same IP, my guess is it’s a Sonicwall. maybe “force” each one to register on a differnt porrt i.e. 5061 for one and 5062 for the othet, some stupid routers make ass-u-mptions about 5060 that they shouldn’t

Hi tadpole,

What do you mean by that ? I am alreadying forcing the 2 connections to 2 differents ports : 5060 and 5061. Plus, I have done an iptables Rule on my SIP server :

Redirecting port 5061 to 5060

-A PREROUTING -p udp -m udp -d 8x.9x.14x.x3x --dport 5061 -j DNAT --to-destination

Do you think it’s coming from my client’s router or sip server’s firewall ?

Thanks !

I have found something !

The attended transfer is working as the blind doesn’t :slight_smile:

There is probably an answer somewhere :wink:

any idea?

What do you think,? 3304 is registering at 58673 and 3300 at 46853

Registered SIP ‘3304’ at 88.x8x.11.xx:58673
Registered SIP ‘3300’ at 88.x8x.11.xx:46853

presumably by your client router

Having asked the phones to register on “non-standrd” ports and then immediately re-translating 5061 to 5060 back on the same address, then there is another problem.

Properly configured routers and NAT = yes will let SIP and asterisk re-write the headers quite correctly.