Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
I had the UDP ports forwarded, In any case I will be testing with a brand new router today, then will confirm if my old router had a problem.On 11/24/05, Tom Rymes [EMAIL PROTECTED] wrote: On Nov 24, 2005, at 12:14 PM, Bharath wrote: I found out that I have a faulty Belkin Router which was causing the problem. I tried forwarding ports as well as DMZ'd the Sip device but still could'nt not hear the voice. So i plugged the sip device directly to the cable modem it worked fine. So my guess is that though I have set up the router to forwards port to the sip device there is something happening at the router that is blocking the RTP ports (1-2). ThanksBefore you blame the router, make sure that you forwarded UDP ports5060 and 1-2, not TCP. (Though I guess the DMZ setup wouldhave taken care of that...) TomTom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414Intelligent technology solutions for small businesses. ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On Nov 25, 2005, at 7:00 PM, Manny A. Wise wrote: -Original Message- From: Tom Rymes [mailto:[EMAIL PROTECTED] [snip] On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote: [snip] Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT.. if you make it work just by opening ports, let me know..I have never been able to get it to work, that's why I don't use sip, just plain iax2 for everything. J Manny Manny, I have this working as I write this. (I just hung up the phone.) In fact, I brought a Cisco 7940G to a completely unknown nat-ed network the other day, plugged it in and started making calls right away. Here's the setup I have for this specific configuration: 1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but it's still NAT. I just don't have to forward ports this way) 2.) externip, localnet, nat settings configured in the sip.conf file (sip_nat.conf for [EMAIL PROTECTED]) 3.) Cisco phone (or whatever SIP UA you choose) configured for NAT (via the SIPMAC.cnf file for Cisco) 4.) Lather, rinse, repeat if necessary Hopefully that will work for you. I'd rather use IAX and avoid these problems altogether, but I have yet to find an IAX hardphone I am willing to use. In fact, for softphone use, I do indeed use IAX via LoudHush for the mac. (Great piece of software, BTW. No connection here, just a happy user...) Tom Great!!, this did the trick, now we have audio... We are using a Sipura 2000 for testing The Sipura now can call out and have audio...the only problem left is that the sipura can't receive calls, when the extension is dialed, the recording says, the person is on the phone.any ideas??? I changed the externip=, localnet= and nat=yes in sip.com and in the extension setup in amp nat=1.. missing anything THANKS Manny It sounds as if your extension isn't registered. Make sure that the extension is configured as dynamic in sip.conf (or AMP) and as nat=yes. Also, make sure that the Sipura is configured through its web interface to register and it has the right user and password entered. Once this is done, when you type 'sip show peers' from the CLI your Sipura's extension should be listed, and show a 'D' and an 'N' for dynamic and nat. Also, it sounds like you are using AMP and or [EMAIL PROTECTED], so make sure that you put the nat, externip, and localnet parameters in the sip_nat.conf file, *NOT* the sip.conf file, as that is likely to get overwritten by AMP. From my installation (obviously, substitute your external IP for the xxx.xxx.xxx.xxx below...): [EMAIL PROTECTED] root]# cat /etc/asterisk/sip_nat.conf nat=yes externip=xxx.xxx.xxx.xxx localnet=10.0.0.0/255.255.255.0 Other than that, I recommend further google and voip-info spelunking expeditions to track down your problem. I think that voxilla.com also has good resources on the Sipuras Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote: [snip] Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have never been able to get it to work, that’s why I don’t use sip, just plain iax2 for everything… J Manny Manny, I have this working as I write this. (I just hung up the phone.) In fact, I brought a Cisco 7940G to a completely unknown nat-ed network the other day, plugged it in and started making calls right away. Here's the setup I have for this specific configuration: 1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but it's still NAT. I just don't have to forward ports this way) 2.) externip, localnet, nat settings configured in the sip.conf file (sip_nat.conf for [EMAIL PROTECTED]) 3.) Cisco phone (or whatever SIP UA you choose) configured for NAT (via the SIPMAC.cnf file for Cisco) 4.) Lather, rinse, repeat if necessary Hopefully that will work for you. I'd rather use IAX and avoid these problems altogether, but I have yet to find an IAX hardphone I am willing to use. In fact, for softphone use, I do indeed use IAX via LoudHush for the mac. (Great piece of software, BTW. No connection here, just a happy user...) Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
I found out that I have a faulty Belkin Router which was causing the problem. I tried forwarding ports as well as DMZ'd the Sip device but still could'nt not hear the voice. So i plugged the sip device directly to the cable modem it worked fine. So my guess is that though I have set up the router to forwards port to the sip device there is something happening at the router that is blocking the RTP ports (1-2). Thanks On 11/24/05, Tom Rymes [EMAIL PROTECTED] wrote: On Nov 23, 2005, at 11:04 AM, Manny A. Wise wrote:[snip] Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have never been able to get it to work, that's why I don't use sip, just plain iax2 for everything… J MannyManny,I have this working as I write this. (I just hung up the phone.) In fact, I brought a Cisco 7940G to a completely unknown nat-ed networkthe other day, plugged it in and started making calls right away.Here's the setup I have for this specific configuration:1.) Asterisk server behind NAT. (Setup as DMZ on Linksys WRT54G, but it's still NAT. I just don't have to forward ports this way)2.) externip, localnet, nat settings configured in the sip.conf file(sip_nat.conf for [EMAIL PROTECTED])3.) Cisco phone (or whatever SIP UA you choose) configured for NAT (via the SIPMAC.cnf file for Cisco)4.) Lather, rinse, repeat if necessaryHopefully that will work for you. I'd rather use IAX and avoid theseproblems altogether, but I have yet to find an IAX hardphone I am willing to use. In fact, for softphone use, I do indeed use IAX viaLoudHush for the mac. (Great piece of software, BTW. No connectionhere, just a happy user...)TomTom Rymes Cascade Link Systemswww.cascadelinksystems.com(603) 375-1414Intelligent technology solutions for small businesses.___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On Nov 24, 2005, at 12:14 PM, Bharath wrote: I found out that I have a faulty Belkin Router which was causing the problem. I tried forwarding ports as well as DMZ'd the Sip device but still could'nt not hear the voice. So i plugged the sip device directly to the cable modem it worked fine. So my guess is that though I have set up the router to forwards port to the sip device there is something happening at the router that is blocking the RTP ports (1-2). Thanks Before you blame the router, make sure that you forwarded UDP ports 5060 and 1-2, not TCP. (Though I guess the DMZ setup would have taken care of that...) Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? ThanksOn 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf : [2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED] host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath KhambadkoneSent: Wednesday, November 23, 2005 9:29 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs?Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All,I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front ofIf no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed.my Sip.conf :[2008] ;(Sipura2002)username=2008type=friendsecret=2008record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED] host=dynamicdtmfmode=rfc2833context=from-internalcanreinvite=nocallerid=device 2008[2009] ;X-Lite Soft Phoneusername=2009type=friend secret=2009record_out=Adhocrecord_in=Adhocqualify=noport=5060nat=1[EMAIL PROTECTED]host=dynamicdtmfmode=rfc2833context=from-internal canreinvite=nocallerid=device 2009Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT. if you make it work just by opening ports, let me know..I have never been able to get it to work, thats why I dont use sip, just plain iax2 for everything J Manny -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 1-2 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West [EMAIL PROTECTED] wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Manny, Sorry if my post caused any confusion. I'm talking about 2 different locations of the server client. My Asterisk server is located at my office and is not behind a NAT or firewall. It is directly connected to my Cable modem. I'm using a Sipura2002 ATA at home. This ATA is connected to the asterisk server which is located at my office. The ATA at my home is behind a NAT. The ATA sucessfully registers and can also make recieve calls only the voice is blocked. The external ports 1-2 were not opened on my Asterisk box. Only port 5060-5082 were opened. I guess thats the reason I was not able to hear any voice. Will try that this evening and post my results. Thanks On 11/23/05, Manny A. Wise [EMAIL PROTECTED] wrote: Well, as the user stated on the original message, the asterisk server is behind a NAT and the client is also behind a NAT…. if you make it work just by opening ports, let me know..I have never been able to get it to work, that's why I don't use sip, just plain iax2 for everything… J Manny -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Bharath Sent: Wednesday, November 23, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Thanks Michael, I think thats is the problem, I have opened only ports 5060-5082, I need to open 1-2 as well. I will try that and post the result when i get back home. Thanks On 11/23/05, Michael West [EMAIL PROTECTED] wrote: I'm pasting something from another user on this list from 14/11/05 I would recommend that you do a little research on google, voip- info.org, and the list archives. To connect to an Asterisk box that sits behind NAT, you need to forward ports 5060 and 1-2 too the asterisk box, and you need to configure the externip, localnet, and nat variables in sip.conf. audio problems are almost always due to the RTP stream (ports 1-2) not being forwarded properly, either due to the port forwarding setup or the sip.conf settings. Tom -- Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Bharath Khambadkone Sent: Wednesday, November 23, 2005 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain By default AMP had NAT=yes in sip.conf, I read in some posts to change it to one, i was just trying my luck if that works. I have tried NAT=yes, The Phone gets registered, I can also make recieve calls but as soon as the call is picked I dont hear anything at both ends. Does this have anything to do with codecs? Thanks On 11/22/05, C F [EMAIL PROTECTED] wrote: On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2)with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
On 11/22/05, Bharath Khambadkone [EMAIL PROTECTED] wrote: Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of If no nat then why do you have nat=1 in sip.conf? the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2) with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2) with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain
change nat=1 to nat=yes From: [EMAIL PROTECTED] on behalf of Bharath Khambadkone Sent: Tue 11/22/2005 12:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP Extension behind NAT,Asterisk on a public domain Hello All, I'm fairly new to asterisk. I have read about the problems about NAT, But can't seem to find a solution. My Asterisk is on a public domain, there is no NAT or firewall in front of the asteris box. I have sucessfully connected iax2 softphones was able to recieve make calls. In the same locations where I have the iax2 extensions working I have set up a a SIP softphone a SIP ATA (Sipura2002). Both teh sip phones are able to register. I can also make recieve calls but cannot hear anything after the call is answered at both ends. I'm not sure what is causing this problem. By the way I'm using SME server 7(centos 4.2) with [EMAIL PROTECTED] installed. my Sip.conf : [2008] ;(Sipura2002) username=2008 type=friend secret=2008 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2008 [2009] ;X-Lite Soft Phone username=2009 type=friend secret=2009 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 2009 Thanks in advance.. winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
