I have 800 kbps in both directions reliably at the endpoint location. When I was testing, there weren't any computers in the office, or any other phones.
The server has a 10 Mb ethernet connection in a datacenter, and I usually don't see more than 8 channels at once, so I don't think it's bandwidth. The endpoints I have been testing on have been rock solid in all other modes of operation, except Pickup. Asterisk is trying to do an external RTP bridge, as evidenced below. How do I make it not do that. I have already specified canreinvite=no for all peers. nat=yes for all the peers except the upstream carriers. It's also set as the global default. Retransmitting #6 (NAT) to (Phone Public IP):1126: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Via: SIP/2.0/UDP (Server IP):5060;branch=z9hG4bK5e02a020;rport From: *88 <sip:[EMAIL PROTECTED] Domain>;tag=as64bce3f7 To: "T & S St. Pierre" <sip:[EMAIL PROTECTED] Domain>;tag=d8b4a9e50086b57 Contact: <sip:[EMAIL PROTECTED] IP> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Communicate Freely 1.4 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) <- Why!!! Content-Type: application/sdp Content-Length: 266 On Tuesday 11 December 2007 00:58, dave cantera wrote: > tim, > sounds like a problem I had with bandwidth... too many devices > communicating on the same network connection to the internet... have you > tcpdump'd or used a bandwidth tool to see what the usage is? nat=yes or > nat=no? should be yes.. > did you change the router between upgrades? > just some random thoughts.. > daveC > > > > Tim St. Pierre wrote: > Hello Folks. > > I'm wondering if anyone has any helpful hints. > > I recently upgraded to 1.4.11, and I'm having problems with pickup, both > directed, and the pickup feature. > > My server is on the public internet, and all phones are behind a NAT > router, somewhere else on the public internet. > > When a ringing phone is picked up by another phone, you have audio for a > few seconds, then the call is dropped. > > The console shows "No response to our critical packet" > > A SIP debug of the conversation between the phone and the server shows a > re-invite request right when the call drops. The phone is of course using > the internal IP address as it's contact, and it looks to me like the server > is trying to use it. > > I have canreinvite=no for both the general sip.conf, as well as per-peer. > > I am using the whole range of Aastra Enterprise IP phones. > > Interestingly enough, some phones show their true IP address and port in > the Asterisk registration database. I believe this is where the phones > have successfully communicated with a uPNP router, and discovered their > public address. These phones can successfully pickup the call. > > If I pipe the pickup call through the Local channel, it works. > > Why is asterisk still trying to re-invite even though I have explicitly > told it not to in the config? > > It worked fine in 1.2 > > Any suggestions, or requests for more information? > > Thanks for any help. > > -Tim -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
