Re: [Asterisk-Users] grandstream asterisk configuration
On Thursday 15 January 2004 01:30 am, Chandra wrote: hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. Why leave a host to defend for itself? At least behind a firewall you got some layers of protection. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
On Thu, Jan 15, 2004 at 02:45:22AM -0500, Steve said: On Thursday 15 January 2004 01:30 am, Chandra wrote: hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. Why leave a host to defend for itself? At least behind a firewall you got some layers of protection. Um, just becasue a host isn't NATed, doesn't meant that there isn't a firewall protecting it. Nat is an evil hack used to allow more hosts to use the internet than you have real IP addresses. NAT itself is NOT a firewall, but it does have a side effect by nature that gives NATed hosts some protection. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable Chandra, I have this _exact_ same problem and it's the Netgear router corrupting the UDP checksums in the RTP packets. Specifically, the checksums come out of the phone unset and the router is setting them to incorrect values. Netgear has not yet responded to my support requets. Ethereal will confirm if you're getting the same thing. Swap out the Netgear with a Linksys or other router and I bet it works. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
On Wednesday 14 January 2004 01:52 pm, Mike Machado wrote: On Wed, 2004-01-14 at 08:45, SW wrote: Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). You don't need stun to work with Grandstream. My * is behind NAT and so is the GS of course. Two ports are open and redirected in the F/W, udp 4569 and 5036. I make and receive internal and external calls over both PSTN and the Internet. GS is configured: Software V 1.0.4.30 Static IP SIP Server is Asterisk's IP SIP user ID is the extension of GS Authenticate ID as user ID No pw Name is Steve Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723 723 Rate is 6.3 Silence Suppression is Yes Voice Frames are 2 IP SoQ is 48 VLAN 0 SIP User is NOT phone number Dial Plan 202 SIP register YEs Clear Reg oin reboot NO Expiration 60 Early Dial No Use # as Dial Key is Yes SIP port 5060 RTP 5004 Random port is No NAT traversal is NO keel alive is 20 TFTP server is 130.94.123.253 Voice mail ID is 78202 DTMF is in-audio Payload is 101 - this may need to be changed NTP time.nist.gov Now all my features used to work a few months ago. I then stopped using * and came back a week ago. Updated CVS and now Hold is not working unless I press #(!?) But I can call, receive, transfer and have a working V/M. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
i just saw a UDP blocked message in my gs GUI. ater i rebooted again i got MAC Address:00.0B.82.00.3C.13 Software Version:Program--1.0.3.81Bootloader--1.0.0.7 HTML--1.0.0.18 detected firewall/NAT type is open Internet assigning a STUN server also didn't help. lloked at the voip-info stuff a.. use dtmfmode=info in your sip.conf for your Grandstream BudgeTone and configure the GS accordingly b.. make sure to have a username=xxx entry in sip.conf that matches the phone's name as given in the square brackets c.. For most installations, this is needed in the sip.conf user definition (not in [general]): disallow=all allow=ulaw allow=alaw and did the same. still didn't work. what can be done if my nat is actually blocking the udp packets?? chandra - Original Message - From: SW [EMAIL PROTECTED] To: Chandra [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 10:30 PM Subject: Re: [Asterisk-Users] grandstream asterisk configuration Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: [EMAIL PROTECTED] i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
hi i am not talking about * behind NAT. its * outside NAT and GS inside NAT. chandra - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 9:50 AM Subject: Re: [Asterisk-Users] grandstream asterisk configuration On Wednesday 14 January 2004 01:52 pm, Mike Machado wrote: On Wed, 2004-01-14 at 08:45, SW wrote: Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). You don't need stun to work with Grandstream. My * is behind NAT and so is the GS of course. Two ports are open and redirected in the F/W, udp 4569 and 5036. I make and receive internal and external calls over both PSTN and the Internet. GS is configured: Software V 1.0.4.30 Static IP SIP Server is Asterisk's IP SIP user ID is the extension of GS Authenticate ID as user ID No pw Name is Steve Codex are: PCMU, PCMA, PCMU, G729, G726, G728 - note no 723 723 Rate is 6.3 Silence Suppression is Yes Voice Frames are 2 IP SoQ is 48 VLAN 0 SIP User is NOT phone number Dial Plan 202 SIP register YEs Clear Reg oin reboot NO Expiration 60 Early Dial No Use # as Dial Key is Yes SIP port 5060 RTP 5004 Random port is No NAT traversal is NO keel alive is 20 TFTP server is 130.94.123.253 Voice mail ID is 78202 DTMF is in-audio Payload is 101 - this may need to be changed NTP time.nist.gov Now all my features used to work a few months ago. I then stopped using * and came back a week ago. Updated CVS and now Hold is not working unless I press #(!?) But I can call, receive, transfer and have a working V/M. -- Steve __ You actually need to constantly be alert and willing to handle things, or life will find a way to get you good! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream asterisk configuration
hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is =i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error:Resource temporarily unavailable my sip.conf file is as follows: [general]port =3D 5060 ; Port to bind tobindaddr =3D 0.0.0.0 ; Address to bind to;externip =3D 200.201.202.203 ; Address that we're going to put in =SIPmessages if we're behind a NATtos=3Dlowdelaydisallow=3Dall ; Disallow all codecsallow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1]type=3Dfriendhost=3Ddynamicsecret=3Dmysecretcontext=3Doutgoingnat=3Dyesreinvite=3Dnocanreinvite=3Dnoqualify=3D2000 has anyone done this before? chandra
Re: [Asterisk-Users] grandstream asterisk configuration
Make sure that udp packets can get from the server back to the grandstream. At 12:40 14/01/04, you wrote: hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: [general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra - Original Message - From: bam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 6:42 PM Subject: Re: [Asterisk-Users] grandstream asterisk configuration Make sure that udp packets can get from the server back to the grandstream. At 12:40 14/01/04, you wrote: hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: [general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
You may need to run Ethereal to make sure packets are REALLY getting through. On Wed, Jan 14, 2004 at 07:35:48PM +0545, Chandra said: i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra - Original Message - From: bam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 6:42 PM Subject: Re: [Asterisk-Users] grandstream asterisk configuration Make sure that udp packets can get from the server back to the grandstream. At 12:40 14/01/04, you wrote: hi, I have the following configuration: Grandstream -- NAT (Netgear RP614)--Internet--Asterisk(public IP) i can register fine and call ringing is working as good. The problem is = i cant hear audio both ways and i get this error: WARNING[22544]: File rtp.c, Line 375 (ast_rtp_read): RTP Read error: Resource temporarily unavailable my sip.conf file is as follows: [general] port =3D 5060 ; Port to bind to bindaddr =3D 0.0.0.0 ; Address to bind to ;externip =3D 200.201.202.203 ; Address that we're going to put in = SIP messages if we're behind a NAT tos=3Dlowdelay disallow=3Dall; Disallow all codecs allow=3Dulaw ; Allow codecs in order of preference dtmfmode=3Dinfo [grandstream1] type=3Dfriend host=3Ddynamic secret=3Dmysecret context=3Doutgoing nat=3Dyes reinvite=3Dno canreinvite=3Dno qualify=3D2000 has anyone done this before? chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: [EMAIL PROTECTED] i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] grandstream asterisk configuration
On my handytone, if I did not enable STUN, the * box would send the RTP data to my 192.168 address, even though I had nat=yes in sip.conf and the SIP handshake happened with my public IP. It seemed * was not properly translating (ignoring) the IP that was in the SDP into the IP that the SIP handshake happened on. When I enabled STUN, * then started to send the RTP to the correct IP and port, but I still did not have audio. That was as far as I got and have not had time to run additional traces to see what the holdup is now. On Wed, 2004-01-14 at 08:45, SW wrote: Hi, In my experience with GS phones, you need STUN support to make it work properly (behind NAT), otherwise you would need lot of trial end error to figure out how to do port forwarding. If you have STUN you wouldn't need to touch the Netgear (except for firewalls). If you can't run your own stun server (need two public IPs) then use one of many STUN servers out there on public internet. For an example enable NAT traversal on your GS phone and point the STUN server to one of these STUN servers larry.gloo.net or stun01.newkinetics.com. Then reboot the GS and see how it discover the NAT (top of the gs web GUI). If it is not a full cone or UDP blocked then you should be fine (Netgear is restricted cone). Cheers SW From: Chandra [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] grandstream asterisk configuration Date: Wed, 14 Jan 2004 19:35:48 +0545 Reply-To: [EMAIL PROTECTED] i have forwarded ports 5060 and 5000-5008 the ports used by sip and rtp to grandstream from my netgear. rtp.conf uses rtpstart 5000 and rtpend 5008. i have also opened all 5060, 5000-5008 ports in my firewall configuration. grandstream uses 5004 port for rtp. what am i missing here? please tell me. chandra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
