Sounds like the pause is just your typical call time-out, most handsets will wait a few seconds before pushing the call to the server, on some models you can hit the dial button to make it go quicker, you can probably customize that time-out but it really depends on if its a setting of the firmware or the config files.
I don’t think that’s the case here. It NEVER happened before and it doesn’t happen all the time. The problem seems like its a connection issue. Likle it’s taking too much time to get ahold of the phone to make the transfer or something. Especially with the error that comes up a lot
[2012-10-31 10:00:52] NOTICE[3281]: chan_sip.c:23253 handle_request_invite: Unable to create/find SIP channel for this INVITE
I’m just looking for reasons that this might be happening. I guess I just need to get the paid support or something.
What happens when you append with the #?
BF
when I use # to transfer the call it goes but it stalls for a bit. It’s like, you hit the # and transfer and look down at the phone display and everything is stuck for about 3 or 4 seconds then it goes. That help? Let me know if you need any code or settings. Thanks a lot.
Was this ever resolved? We seem to be encountering identical issue.
thx
I hate to sound like a broken record but: This a NAT problem, it’s always a NAT problem.
Asterisk is very picky about making sure it has RTP continuity. If any ports are not coming back where they are supposed to be it’s going to tear down the call, period. This is what it should do because it improves the user experience by not allowing bad calls to process.
In the Cisco environment make sure you have SIP FIXUP off. They may have renamed it in the latest Security IOS release.
Make sure your VPN subnets are declared in SIP localnet.
It is imperative to understand that the ‘localnet’ Asterisk parameter is poorly named for people that don’t understand networking. A local network is defined as a network that has a connected route to a gateway on the same collision domain (the network defined by the IP address and subnet mask) as the host (your Asterisk box).
An easier way to think of it is networks declared in the localnet field are excluded from NAT processing (application of external IP parameters).
So bottom line is if you have VPN’s, MPLS, point to point t1’s, metro Ethenet, Layer 2 fiber, microwave, lighspan…you get the idea, those networks should be routed and not have any NAT translations. If your interfaces are in NAT mode on your routers, fix it!
Keep in mind this is complicated stuff, it requires an in depth understanding of networking (at least CCNP level), VoIP and Asterisk specifically. You are not going to pick it up reading a few messages on a forum.
VPN’s and other Layer 2 products are now available in very affordable packages from the providers and hardware. The problem with it is the complexity of deploying a multiservice WAN has not decreased, it’s a serious undertaking.
BTW the rate for FreePBX services on networks and other non Asterisk/FreePBX work is upcharged depending on the consultant that needs to be engaged.