[asterisk-users] canreinvite problems

[Stefan van der Eijk]

Hi,

I've been working on migrating my asterisk from zap to sip (due to
compatibility issues between my TDM400P and my Hauppauge PVR500). I've
purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP
phone). I managed to get it all working with my asterisk 1.4.0 installation,
but I'm seeing some interesting things with the canreinvite option that I
can't explain, even after reading:
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite

My setup:

   - asterisk server with:
      - eth0 = 192.168.254.254 (internal network)
      - eth1 = Internet IP-address
      - ZAP/1 (FXO) not used
      - ZAP/2 (FXS) not used
      - ZAP/3 and ZAP/4 (FXS) with DECT phones
      - SPA-3102:
   - WAN interface configured with DHCP, it gets
192.168.254.104(internal network)
      - LAN interface is not being used
      - Line1: DECT phone
      - PSTN: is connected to the PSTN
      - Siemens SL75 WLAN: 192.168.254.105
   - Laptop (192.168.254.125) with an Eyebeam and idefisk softphone

All the SIP endpoints are connected to the internal network, there should be
no NAT issues.

In all situations I'm able to dial the other phone and make it ring.

>From the ZAP endpoints to the SIP endpoints (and vice versa) I get sound.
Same applies to the IAX2 client (idefisk).

When I have 2 SIP endpoints that both aren't configured with
"canreinvite=no" then I get no sound.

Conclusion: all media needs to go through the asterisk server in order to
get sound.

Questions:

   1. Are all of my SIP endpoints incompatible with the canreinvite=yes
   option?
   2. Is there a list of SIP endpoints that are known to work with
   "canreinvite=yes"?
   3. Are other people also experiencing this?

with kind regards,

Stefan van der Eijk
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