Re: [asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio
On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote: Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup: asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? You are aware that SIP only sets up, monitors and takes the call down? The audio stream is RDP and on higher ports. My guess is that the audio stream on inbound calls is not arriving where it should be - or is blocked. This could be router or nat, but one thing jumps out to me: Does your Asterisk Server itself have something set up in the built in iptables firewall blocking udp inbound traffic in the port range 15000:2? The output of the command 'iptables -nvL' will tell you pretty quickly. HTH. -- _ -- 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
Re: [asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio
thanks brian, yes, i am aware that sip is only responsible for signalling and therefor my conclusion was, that it has got something to do with nat / firewall / the router... meanwhile i´ve got it solved... although the sip-provider tried to convince me, that the misconfiguration is on my asterisks´ side, i penetrated the support until they looked over it again and... what should i say... finally they had to admit, that the router had a wrong acesslist. they corrected it and now it works. yves Brian schrieb: On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote: Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup: asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? You are aware that SIP only sets up, monitors and takes the call down? The audio stream is RDP and on higher ports. My guess is that the audio stream on inbound calls is not arriving where it should be - or is blocked. This could be router or nat, but one thing jumps out to me: Does your Asterisk Server itself have something set up in the built in iptables firewall blocking udp inbound traffic in the port range 15000:2? The output of the command 'iptables -nvL' will tell you pretty quickly. HTH. -- _ -- 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
[asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio
Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup: asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? thanks, yves -- _ -- 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
