Re: [asterisk-users] Multiple Asterisk SIP Server/client connections
Hi Ken - The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. I just want to double-check the setup you have: you say the main server has the same config for each remote site connecting. Does that mean they're all connecting to the same SIP user/friend account? If so, that wouldn't work. You need to have a unique SIP account for each SIP device that's connecting. If that's not the case, and you have a unique sip account for each of your Polycom devices, can you show us the relevant part of your sip.conf from the main asterisk server? Also, do you get any particular messages on the console regarding this? Have you tried turning on SIP debugging? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk SIP Server/client connections
When I said same config I meant same with minor differences of account information :D [103] type=friend secret=1234 dial=SIP/103 callerid=Video103 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite [104] type=friend secret=1234 dial=SIP/104 callerid=Video104 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite I get no errors on the main server side, I get a lot of retransmitting messages on the remotes that don't get a connection. I'll post a portion of the log file later. The strange part to me is that it seems as if it's the first person in that wins. I can shutdown remote servers and bring them up individually and they work, but after that initial one it's as if the main server's ignoring port 5060 from other locations. I thought perhaps it was a firewall/router problem on the main server, so I swapped out a netgear router for a linksys wrt54g, same problem occurs on both routers. All 4 servers are listed as DMZ on their local firewalls. Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, July 30, 2008 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple Asterisk SIP Server/client connections Hi Ken - The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. I just want to double-check the setup you have: you say the main server has the same config for each remote site connecting. Does that mean they're all connecting to the same SIP user/friend account? If so, that wouldn't work. You need to have a unique SIP account for each SIP device that's connecting. If that's not the case, and you have a unique sip account for each of your Polycom devices, can you show us the relevant part of your sip.conf from the main asterisk server? Also, do you get any particular messages on the console regarding this? Have you tried turning on SIP debugging? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple Asterisk SIP Server/client connections
I have 4 asterisk servers. They all have local phones on their local network they manage for SIP based conversations. We then have IAX between them all for inter-asterisk connections. This setup has worked well for nearly 2 years now, minor problems here and there but overall very nice. Recently we acquired some Polycom video conference units. I was able to setup our main server to host all the video coordination using video over SIP. I was able to configure the video conference units on the local network, have all 4 of them (one going to each remote server) displaying 4 videos on the local network. I then sent them out to their remote facilities and setup Asterisk with as a SIP client on the 3 remote locations to talk to the main server. One at a time we tested them and they worked one on one. Recently we tried to get two going, and I noticed there seems to be an issue with the SIP registration if one of the 3 remote SIP clients has already registered. That is, the other requests are unanswered or not fully registered for some reason or another. At very random times I've actually managed to get 2 of the 3 connected, but inevitably I lose one of those 2 shortly after. The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. Assuming it is possible, here is the SIP Client SIP.CONF: [general] register = 103:[EMAIL PROTECTED]/699 defaultexpirey=1800 maxexpirey=3600 relaxdtmf=yes videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no limitonpeer=yes notifyringing=yes notifyhold=yes externip=xx.xx.xx.xx.xx fromdomain=xx.xx.xx.xx localnet=192.168.0.0/255.255.255.0 [yy.yy.yy.yy] type=friend host=yy.yy.yy.yy insecure=port,invite [699] type=friend secret=1234 dial=SIP/699 callerid=Video 699 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite In addition here's the relevant portions of the SIP.CONF from the main server: [general] videosupport=yes disallow=all allow=ulaw allow=gsm allow=h263p canreinvite=no fromdomain=yy.yy.yy.yy externip=yy.yy.yy.yy localnet=10.200.26.0/255.255.255.0 nat=yes bindport=5060 [103] type=friend secret=1234 dial=SIP/103 callerid=Video103 allowsubscribe=yes host=dynamic context=from-internal insecure=port,invite Please, any suggestions would be great. I've been bashing my head against the keyboard all day trying to find why it's acting in this way. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
