Re: [asterisk-users] Curious problem with NAT
Zitat von Steve Totaro : Not without seeing some SIP debug output. I'm currently not at home. If you say me which debug output you wish, I can send them as soon I'll be back... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello wrote: > Zitat von Steve Totaro : > > Are you using the wifi on on the cellphone? The peer IP is showing as >> 192.168.200.3 which is not a routable address. Unless things have >> changed, >> double NAT configurations do not work. >> > > Hi Steve, > > My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct > in Internet. > But maybe my Provider does a NAT, too... > > Very strange is, that I have a very poorly audio-quality, if I use my > cellphone in my WLAN and connect to my Asterisk. > With THE SAME USER, but from a PC in the same Network, the audio quality > is perfect. > > Any idea? > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) > > > > Not without seeing some SIP debug output. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Very strange is, that I have a very poorly audio-quality, if I use my cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio quality is perfect. Any idea? Did you check which codecs are active? What does "sip show channelstats" say? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Zitat von Steve Totaro : Are you using the wifi on on the cellphone? The peer IP is showing as 192.168.200.3 which is not a routable address. Unless things have changed, double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a NAT, too... Very strange is, that I have a very poorly audio-quality, if I use my cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio quality is perfect. Any idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
On Sun, Jun 7, 2015 at 8:46 AM, Steve Totaro wrote: > > > On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello > wrote: > >> Ashwin Surendran schrieb: >> >> > What settings have you got for directmedia? >> > >> > Could you try >> > >> > nat=force_rport,comedia >> > directmedia=no >> >> Tried. Peer always unreachable, call not possible... :( >> >> Other idea? >> >> Thanks >> Luca Bertoncello >> (lucab...@lucabert.de) >> >> > > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. > > Thanks, > Steve T > You could try using your carrier's internet access instead of wifi. OpenVPN for Android looks like it could work to eliminate your NAT issues as well. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
On Sun, Jun 7, 2015 at 4:49 AM, Luca Bertoncello wrote: > Ashwin Surendran schrieb: > > > What settings have you got for directmedia? > > > > Could you try > > > > nat=force_rport,comedia > > directmedia=no > > Tried. Peer always unreachable, call not possible... :( > > Other idea? > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) > > Are you using the wifi on on the cellphone? The peer IP is showing as 192.168.200.3 which is not a routable address. Unless things have changed, double NAT configurations do not work. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Ashwin Surendran schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
What settings have you got for directmedia? Could you try nat=force_rport,comedia directmedia=no -Ashwin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 12:04 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Curious problem with NAT Ashwin Surendran mailto:ashwin.surend...@now-health.com>> schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucab...@lucabert.de<mailto:lucab...@lucabert.de>) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Ashwin Surendran schrieb: > Have you tried NAT=force_rport ? OK, tried... I can transmit from my phone (aka: I hear my voice on another phone), but I'm not able to receive data (aka: I cannot hear what I say on the other phone). Other suggestion? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Ashwin Surendran schrieb: > Have you tried NAT=force_rport ? No, not yet... I'll try later and report to the list... Have I to define (in Asterisk or Gateway) the ports? Thanks Luca bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Curious problem with NAT
Have you tried NAT=force_rport ? Ashwin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luca Bertoncello Sent: 07 June 2015 11:44 To: Asterisk Users Subject: [asterisk-users] Curious problem with NAT Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [0049177111] fullname = 0049177111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/0049177111 and finally "core reload". On my Gateway I configured the NAT so: /sbin/iptables -t nat -A PREROUTING -p udp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p udp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 Well, the phone connect to the server and I can see it reachable: OpenWrt*CLI> sip show peers Name/username HostDyn Forcerport ACL Port Status 0049177111/0049177111 192.168.200.3D N 40702OK (1768 ms) Well, now I call the mobile phone from another peer. It rings and I can answer the call. Wonderful! But no word will be sent... :( I cannot hear anything on my mobile phone and I cannot transmit a single word I tried to connect my mobile phone to a public VoIP-Provider and it works as expected, so I'm sure that the problem is on my network, but I can't find it... What am I doing wrong? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Curious problem with NAT
Hi list! Since the internal calls work as expected and I can register my Asterisk on an external provider, I'd like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls. Well, first of all, my Asterisk is NOT direct on Internet available, but behind a NAT. So I configured my sip.conf: localnet=192.168.200.0/24 externhost=myhost.noip.com externrefresh=180 Then I added the peer in my users.con: [0049177111] fullname = 0049177111 secret = MYVERYSECRET type=peer nat=yes qualify=yes qualifyfreq=60 hassip = yes dahdichan = 1 transport=udp,tcp callwaiting = no context = default host = dynamic dtmfmode=rfc2833 dial=SIP/0049177111 and finally "core reload". On my Gateway I configured the NAT so: /sbin/iptables -t nat -A PREROUTING -p udp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --sport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p udp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 /sbin/iptables -t nat -A PREROUTING -p tcp --dport 6060 -j DNAT --to-destination 192.168.200.120:5060 Well, the phone connect to the server and I can see it reachable: OpenWrt*CLI> sip show peers Name/username HostDyn Forcerport ACL Port Status 0049177111/0049177111 192.168.200.3D N 40702OK (1768 ms) Well, now I call the mobile phone from another peer. It rings and I can answer the call. Wonderful! But no word will be sent... :( I cannot hear anything on my mobile phone and I cannot transmit a single word I tried to connect my mobile phone to a public VoIP-Provider and it works as expected, so I'm sure that the problem is on my network, but I can't find it... What am I doing wrong? Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
