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
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
Are your phones behind NAT? This should be an issue with rtp port communication. Gerard. ------Original Message------ From: John Koenig Sender: asterisk-users-bounces at lists.digium.com To: asterisk-users at 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
I had issues like this on one installation that cleared up when I turned ACPI and APIC?? off in bios. Darren Wiebe darren at aleph-com.net 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: asterisk-users-bounces at lists.digium.com > To: asterisk-users at 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 >
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: asterisk-users-bounces at lists.digium.com > To: asterisk-users at 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 >-------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: sip.conf Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20080715/1e4e9a0a/attachment.txt