[Asterisk-Users] Video Support Not Working
Title: Video Support Not Working Hello, I have a couple of video phones that I am trying to get setup. I have used these phones with sipphone.com and they work great. Now I am trying to get them to work with my * server and I am having problems. The voice portion seems to work fine, but I can not get video to work. These phones and my * server are all on the same subnet connected to the same Ethernet switch. I have included my sip.conf below for reference. I also get a message in the * console each time I establish a call: WARNING[9298]: chan_sip.c:6134 receive_info: Unable to parse INFO message from [EMAIL PROTECTED] Content. Any help that anyone can provide is very much appreciated. Thank you, Nathan [general] context=sip port=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=g729 allow=h263 allow=h261 allow=alaw allow=ulaw allow=iLBC musicclass=default language=en rtptimeout=60 rtpholdtimeout=300 useragent=Asterisk PBX promiscredir = yes videosupport=yes [101] type=friend context=sip username=101 secret=101 fromuser=101 callerid=101 101 host=dynamic nat=no canreinvite=yes qualify=200 dtmfmode=rfc2833 [201] type=friend context=sip username=201 secret=201 fromuser=201 callerid=201 201 host=dynamic nat=no canreinvite=yes qualify=200 dtmfmode=rfc2833 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Starting RC1
Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service asterisk start'. I get the following errors logged to /var/log/asterisk/messages each time I try to start: Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already registered (or something close enough) Jul 19 17:32:26 WARNING[1076227072]: Already have an application 'ParkedCall' Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, returning -1 Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, 'res_parking.so' has use count 1 Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so failed! Any ideas would be great. Thank you, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem Starting RC1
This worked great! Thank you, Nathan -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 5:58 PM To: Nathan Martinez Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem Starting RC1 I had the same problem. Before you make install from the asterisk directory, try removing all the files in /usr/lib/asterisk/modules . That should resolve any potential conflicts from stuff left over from the last build. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 19 Jul 2004, Nathan Martinez wrote: Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service asterisk start'. I get the following errors logged to /var/log/asterisk/messages each time I try to start: Jul 19 17:32:26 WARNING[1076227072]: Command 'showparkedcalls' already registered (or something close enough) Jul 19 17:32:26 WARNING[1076227072]: Already have an application 'ParkedCall' Jul 19 17:32:26 WARNING[1076227072]: res_parking.so: load_module failed, returning -1 Jul 19 17:32:26 WARNING[1076227072]: Soft unload failed, 'res_parking.so' has use count 1 Jul 19 17:32:26 WARNING[1076227072]: Loading module res_parking.so failed! Any ideas would be great. Thank you, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream HT-286 and NAT
I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the DMZ Host in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 == NAT FW == * Server === Router/FW == HT-286-2 In the setup for HT-286-2 , I have filled in the Use NAT IP field with the public IP for that location. I did the same thing for HT-286-1 and then I mapped a public IP to its private IP in the NAT FW. At this point, the two devices can call each other without any problems. I want to use the HT-286 for our traveling users who will never know what their IP is. When I remove the Use NAT IP entry on HT-286-1 as well as remove its direct IP mapping from the NAT FW, HT-286-1 can register with the * Server, but when I try to call HT-286-2, all I get is silence. If I do a 'sip show channels' it shows that the call is connected. Here is what I have in my sip.conf for these two units: [305] type=friend host=dynamic nat=yes qualify=100 [307] type=friend host=dynamic nat=yes qualify=100 Has anyone used these units in this scenario? Does anyone have any hints as to what I can try to get this working? Your help is much appreciated. Nathan Martinez ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
