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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070210/fb0faa3a/attachment.htm
Stefan,> When I have 2 SIP endpoints that both aren't configured with > "canreinvite=no" then I get no sound.The Sipura 3102 definitely works fine with canreinvite=yes and I never really had a problem with any of the Sipura devices in this respect, especially when there is no NAT involved. However, the default "Auto NetService Private IP Ranges:" includes 192.168.0.0-192.168.255.255, so your 192.168.254.0/24 network would be considered a LAN address by the 3102 and hence the traffic would go out the LAN interface (not WAN). Change this setting by removing this range. It's on the Admin > Advanced > LAN Setup tab. If that doesn't help, then you need to check what traffic is being sent. Since all devices are on the same internal network I assume they can see each other. You need to look at the Invite (and ReInvite) messages sent and received and see if the IP addresses for RTP listed there make sense. Then I suggest you use tcpdump to see what traffic is sent by each device, and where. If you have a switched network environment this will be a bit tricky as your * box won't see this traffic, so you may want to use a hub for this test (just temporarily) or if available set up port mirroring to sniff the traffic. Good luck and keep us posted. --Luki