Hi guys, After hours of googling and trying every different config options I need some help! My scenario is as follows: Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun) Main Network (where Freeswitch resides): 10.0.0.0/24 My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34 I can fully use the VPN from both networks (ssh, web gui control, etc) phones can register, however here are my problems: When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up. Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable. I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be). I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box. I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :) Thanks in advance guys!
Hi, Is it only during voicemal? If so, Try this: http://wiki.freeswitch.org/wiki/Variable_record_waste_resources Otherwise turn on sip trace with "sofia global siptrace on" and "console loglevel debug" and post it. On Thu, Jul 18, 2013 at 10:54 AM, Paul <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thanks for the suggestion Anthony, I will check the logs again to see if there is any TIMEOUT errors... (I don't think there was). On Thu, July 18, 2013 7:15 am, Anthony Minessale wrote:
Hi,
Is it only during voicemal? If so, Try this:
http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
Otherwise turn on sip trace with "sofia global siptrace on" and "console loglevel debug" and post it.
On Thu, Jul 18, 2013 at 10:54 AM, Paul <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_______________________________________________________________________ __ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Paul, In my experience there 's only one thing worse than trying to solve your own NAT/double-NAT/ALG issues, and that's trying to solve somebody else's.... But, in all seriousness, I use a similar setup to you, multiple FS instances sitting behind Kamailio as proxy/load balancer and have had no NAT issues after reading "FreeSWITCH 1.2, Chapter 12 - Handling NAT". Alas, particularly without any protocol traces, there are no quick answers here, but like any good diagnostic process/engineer I would suggest you change one parameter at a time & retest, recording the results & moving forward/back a step until you find a configuration that works for you. Seriously, download the book, worth the cover price alone for the NAT chapter & you'll not only be able to solve your problem but will have a greater understanding of the issue(s) and how to plan to avoid them in the future. Rgds, Mick ________________________________ From: Paul <pasha@prosperity4ever.com> To: freeswitch-users@lists.freeswitch.org Sent: Thursday, 18 July 2013, 16:54 Subject: [Freeswitch-users] 30 Second call drop. Hi guys, After hours of googling and trying every different config options I need some help! My scenario is as follows: Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun) Main Network (where Freeswitch resides): 10.0.0.0/24 My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34 I can fully use the VPN from both networks (ssh, web gui control, etc) phones can register, however here are my problems: When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up. Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable. I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be). I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box. I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :) Thanks in advance guys! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Mick, I hear ya on trying to figure out someone else's NAT issues ... I have this weird feeling that it's not NAT though... Only reason I threw my problem out there is just in case someone sees the post and says "Oh yeah, been there, here's what worked for me". I did buy the book on your recommendation so I will be reading through chapter 12 tonight, see if I can figure this out. Thanks for your input! Paul On Thu, July 18, 2013 7:37 am, Mick Stevens wrote:
Hi Paul,
In my experience there 's only one thing worse than trying to solve your own NAT/double-NAT/ALG issues, and that's trying to solve somebody else's....
But, in all seriousness, I use a similar setup to you, multiple FS instances sitting behind Kamailio as proxy/load balancer and have had no NAT issues after reading "FreeSWITCH 1.2, Chapter 12 - Handling NAT".
Alas, particularly without any protocol traces, there are no quick answers here, but like any good diagnostic process/engineer I would suggest you change one parameter at a time & retest, recording the results & moving forward/back a step until you find a configuration that works for you.
Seriously, download the book, worth the cover price alone for the NAT chapter & you'll not only be able to solve your problem but will have a greater understanding of the issue(s) and how to plan to avoid them in the future.
Rgds, Mick
________________________________ From: Paul <pasha@prosperity4ever.com> To: freeswitch-users@lists.freeswitch.org Sent: Thursday, 18 July 2013, 16:54 Subject: [Freeswitch-users] 30 Second call drop.
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I have found in the past that a 30 second call drop is normally a firewall issue. If you have bypass media on: <action application="set" data="bypass_media=true"/> Then at almost exactly 30 seconds FS will take you out of the media path. If your 2 endpoints cannot see each other due to a firewall (or nat) then the media will drop. In our case it's been firewall settings every time. Hope that helps. Tim -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Paul Sent: Thursday, July 18, 2013 11:54 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] 30 Second call drop. Hi guys, After hours of googling and trying every different config options I need some help! My scenario is as follows: Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun) Main Network (where Freeswitch resides): 10.0.0.0/24 My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34 I can fully use the VPN from both networks (ssh, web gui control, etc) phones can register, however here are my problems: When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up. Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable. I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be). I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box. I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :) Thanks in advance guys! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks Tim! I'm not convinced it's firewall because of the VPN, and I allow and masquerade all traffic to and from the local and vpn subnets. All my Asterisk boxes work fine, and so does NGCP, I'm just having this issue with FreeSwitch.. most likely some noob mistake as this is my first attempt at freeswitch :) Paul On Thu, July 18, 2013 10:29 am, Tim Meade wrote:
I have found in the past that a 30 second call drop is normally a firewall issue.
If you have bypass media on:
<action application="set" data="bypass_media=true"/>
Then at almost exactly 30 seconds FS will take you out of the media path. If your 2 endpoints cannot see each other due to a firewall (or nat) then the media will drop.
In our case it's been firewall settings every time.
Hope that helps.
Tim
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Paul Sent: Thursday, July 18, 2013 11:54 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] 30 Second call drop.
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Paul, Had similar issues myself few months back, it ended up port blocked on external firewall. To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response. Sent from my iPhone On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
For some wired reason my last mail is not showing any text, It was "Hi Paul, Had similar issues myself few months back, it ended up port blocked on external firewall. To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response." Don't know my emails are going blank :( Sent from my iPhone On 18 Jul 2013, at 20:55, Kaleem <k4kaleem@gmail.com> wrote:
Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response.
Sent from my iPhone
On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
You messages are going to the list, the list just doesn't send them to you unless you specifically request it to do so... Also, if you are a new user on the list, you may have been automatically pushed into the moderate queue waiting for a list admin to clear you for posting. (an unfortunately nessecary evil to keep the spammers away) On 7/18/13 3:06 PM, "Kaleem" <k4kaleem@gmail.com> wrote:
For some wired reason my last mail is not showing any text,
It was "Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response."
Don't know my emails are going blank :(
Sent from my iPhone
On 18 Jul 2013, at 20:55, Kaleem <k4kaleem@gmail.com> wrote:
Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response.
Sent from my iPhone
On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch
Hi Kaleem, I will double check those settings. I have my firewall setup very specific (it's not wide open to the world).. I h accept port 5060 UDP and all the RTP ports but only if the source IP is of my SIP Trunk provider proxy, otherwise it's invisible. Thanks for the tip! Paul On Thu, July 18, 2013 1:06 pm, Kaleem wrote:
For some wired reason my last mail is not showing any text,
It was "Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response."
Don't know my emails are going blank :(
Sent from my iPhone
On 18 Jul 2013, at 20:55, Kaleem <k4kaleem@gmail.com> wrote:
Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response.
Sent from my iPhone
On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, sometime ago i had something similar...with some old firmware versions of yealink phones. I managed to solved by disabling session_timers in the profile, param, enable_timer=false . hope it helps. António On Fri, 2013-07-19 at 07:49 -0700, Paul wrote:
Hi Kaleem,
I will double check those settings.
I have my firewall setup very specific (it's not wide open to the world).. I h accept port 5060 UDP and all the RTP ports but only if the source IP is of my SIP Trunk provider proxy, otherwise it's invisible.
Thanks for the tip!
Paul
On Thu, July 18, 2013 1:06 pm, Kaleem wrote:
For some wired reason my last mail is not showing any text,
It was "Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response."
Don't know my emails are going blank :(
Sent from my iPhone
On 18 Jul 2013, at 20:55, Kaleem <k4kaleem@gmail.com> wrote:
Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response.
Sent from my iPhone
On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Paul, As u mention port 5060, can u also check for 5080 in your firewall logs and sip traces, Just as curiously I would allow 5080 and see if it helps then remove Sent from my iPhone On 19 Jul 2013, at 15:49, "Paul" <pasha@prosperity4ever.com> wrote:
Hi Kaleem,
I will double check those settings.
I have my firewall setup very specific (it's not wide open to the world).. I h accept port 5060 UDP and all the RTP ports but only if the source IP is of my SIP Trunk provider proxy, otherwise it's invisible.
Thanks for the tip!
Paul
On Thu, July 18, 2013 1:06 pm, Kaleem wrote:
For some wired reason my last mail is not showing any text,
It was "Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response."
Don't know my emails are going blank :(
Sent from my iPhone
On 18 Jul 2013, at 20:55, Kaleem <k4kaleem@gmail.com> wrote:
Hi Paul,
Had similar issues myself few months back, it ended up port blocked on external firewall.
To confirm if its same issue, please test inbound and outbound call, outbound was disconnecting after 30 sec when free switch was sending 200 ok and not receiving response.
Sent from my iPhone
On 18 Jul 2013, at 16:54, "Paul" <pasha@prosperity4ever.com> wrote:
Hi guys,
After hours of googling and trying every different config options I need some help!
My scenario is as follows:
Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun)
Main Network (where Freeswitch resides): 10.0.0.0/24
My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34
I can fully use the VPN from both networks (ssh, web gui control, etc)
phones can register, however here are my problems:
When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up.
Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable.
I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be).
I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box.
I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :)
Thanks in advance guys!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (7)
-
Anthony Minessale
-
Antonio Silva
-
Kaleem
-
Ken Rice
-
Mick Stevens
-
Paul
-
Tim Meade