Re: [asterisk-users] Beginner Issues
Hi John - That could be...I only have ports 5060 and 8088 open on the firewall. Should another port be open? If asterisk is inside a firewall/nat and the phone devices are on the other side, you need to also open port for the rtp audio stream. By default, this is UDP 1 - 2, but this range can be modified in rtp.conf The phone I am using are pstn phones connected to a 2 port Linksys PAP2. I made sure that I checked the NAT option under the user account and enabled NAT Keep Alive under the PAP2 management interface. I am using the G726-16 codec for transmission. Aha. You're using the GUI. In that case, the useful info will be in users.conf. - 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] Beginner Issues
Thanks! Opening the ports did the trick! John Noah Miller wrote: Hi John - That could be...I only have ports 5060 and 8088 open on the firewall. Should another port be open? If asterisk is inside a firewall/nat and the phone devices are on the other side, you need to also open port for the rtp audio stream. By default, this is UDP 1 - 2, but this range can be modified in rtp.conf The phone I am using are pstn phones connected to a 2 port Linksys PAP2. I made sure that I checked the NAT option under the user account and enabled NAT Keep Alive under the PAP2 management interface. I am using the G726-16 codec for transmission. Aha. You're using the GUI. In that case, the useful info will be in users.conf. - 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] Beginner Issues
I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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] Beginner Issues
Hi John - I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? Most probably it's a codec issue, but we'll need to see your sip.conf file. It might also be helpful to know what SIP devices you're using. - 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] Beginner Issues
Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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] Beginner Issues
I had issues like this on one installation that cleared up when I turned ACPI and APIC?? off in bios. Darren Wiebe [EMAIL PROTECTED] Gerard A. Matthew wrote: Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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
Re: [asterisk-users] Beginner Issues
That could be...I only have ports 5060 and 8088 open on the firewall. Should another port be open? The phone I am using are pstn phones connected to a 2 port Linksys PAP2. I made sure that I checked the NAT option under the user account and enabled NAT Keep Alive under the PAP2 management interface. I am using the G726-16 codec for transmission. Attached is my sip.conf. John Gerard A. Matthew wrote: Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. --Original Message-- From: John Koenig Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Jul 15, 2008 6:47 PM Subject: [asterisk-users] Beginner Issues I am new to asterisk, and I am having some troubles. I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? John Koenig ___ -- 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 Sent from my T-Mobile BlackBerry Handheld ___ -- 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 ; ; SIP Configuration example for Asterisk ; ; Syntax for specifying a SIP device in extensions.conf is ; SIP/devicename where devicename is defined in a section below. ; ; You may also use ; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet ; (Don't forget to enable DNS SRV records if you want to use this) ; ; If you define a SIP proxy as a peer below, you may call ; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED] ; where the proxyhostname is defined in a section below ; ; Useful CLI commands to check peers/users: ; sip show peers Show all SIP peers (including friends) ; sip show users Show all SIP users (including friends) ; sip show registry Show status of hosts we register with ; ; sip debug Show all SIP messages ; ; reload chan_sip.so Reload configuration file ; Active SIP peers will not be reconfigured ; [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk. If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ;domain=mydomain.tld; Set default domain for this host ; If configured, Asterisk will only allow ; INVITE and REFER to non-local domains ; Use sip show domains to list local domains ;pedantic=yes ; Enable checking of tags in headers
