Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rules. I tried the "nat=yes" parameter but no changes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091222/915ea9bc/attachment.htm
It is a nat problem Fran?ois BERGANZ P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. De : asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de zehra yildiz Envoy? : mardi 22 d?cembre 2009 10:26 ? : asterisk-users at lists.digium.com Objet : [asterisk-users] asterisk & x-lite Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rules. I tried the "nat=yes" parameter but no changes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091222/e1c29f98/attachment.htm
Try tcpdump to see where RTP go from asterisk. Configure your x-lite Use stun server ? P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. De : asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de zehra yildiz Envoy? : mardi 22 d?cembre 2009 10:26 ? : asterisk-users at lists.digium.com Objet : [asterisk-users] asterisk & x-lite Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend context=phones host=dynamic [root at localhost asterisk]# cat extensions.conf [globals] [general] autofallthrough=yes [default] [incoming_calls] [phones] exten => _1XXX,1,NoOp() exten => _1XXX,n,Dial(SIP/${EXTEN},30) exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) exten => _1XXX,n,Hangup() PS: My sip server and softphones are in the same network subnet. There are not any firewall or iptables rules. I tried the "nat=yes" parameter but no changes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091222/a9fc13f8/attachment.htm
Where is your definition of codecs ?? On Tue, 2009-12-22 at 11:26 +0200, zehra yildiz wrote:> Hello All, > > I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. > The softphone can call the other one but I can' t hear any voice. My > configuration files are below: > > [root at localhost asterisk]# cat sip.conf > [general] > canreinvite=yes > > [1001] > username=1001 > password=1001 > type=friend > context=phones > host=dynamic > > [1002] > callerid=1002 > username=1002 > password=1002 > type=friend > context=phones > host=dynamic > > [root at localhost asterisk]# cat extensions.conf > [globals] > > [general] > autofallthrough=yes > > [default] > > [incoming_calls] > > [phones] > exten => _1XXX,1,NoOp() > exten => _1XXX,n,Dial(SIP/${EXTEN},30) > exten => > _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) > exten => _1XXX,n,Hangup() > > > PS: My sip server and softphones are in the same network subnet. There > are not any firewall or iptables rules. I tried the "nat=yes" > parameter but no changes. > > _______________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091222/58ee6d39/attachment.htm
serach the option en sip.conf: externip = you public ip localnet=tus direcciones locales (address local) saludos Roman On Tue, Dec 22, 2009 at 4:26 AM, zehra yildiz <zyildirim at gmail.com> wrote:> Hello All, > > I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The > softphone can call the other one but I can' t hear any voice. My > configuration files are below: > > [root at localhost asterisk]# cat sip.conf > [general] > canreinvite=yes > > [1001] > username=1001 > password=1001 > type=friend > context=phones > host=dynamic > > [1002] > callerid=1002 > username=1002 > password=1002 > type=friend > context=phones > host=dynamic > > [root at localhost asterisk]# cat extensions.conf > [globals] > > [general] > autofallthrough=yes > > [default] > > [incoming_calls] > > [phones] > exten => _1XXX,1,NoOp() > exten => _1XXX,n,Dial(SIP/${EXTEN},30) > exten => _1XXX,n,Playback(the-party-you-are-calling&is-curntly-unavail) > exten => _1XXX,n,Hangup() > > > PS: My sip server and softphones are in the same network subnet. There are > not any firewall or iptables rules. I tried the "nat=yes" parameter but no > changes. > > _______________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091222/c31f82c8/attachment.htm