SIP Header Via - IP Wrong
My FreeSWITCH server has 2 IP addresses An Internal 192.168,nnn.nn & an External 107,1.nnn.nn This is accomplished via network 'magic' (I'm not the network guy) using a single network card And, in general, it works. I can Receive Outside calls from our SIP Trunk vendor But I can Not Send calls Out through the SIP Trunk vendor. The calls are being blocked (403 'Forbidden") by the vendor due to the contents of our SIP Header packet. Within the Via: part of the packet it shows: Via: SIP/2.0/UDP 192.168.nnn.nn;rport;branch=z9hG4bKQKa7126jtX4mm They need it to show the External IP here. I have even tried setting sip_via_host = 107.1.nnn.nn but that did not change what it being sent. How/Where can I get this to change?
A profile has to listen on a specific IP address. If you want to listen on both IPs you'll need a profile on each one. It's probably easiest in your setup to put the IP addresses into the profile directly rather than setting them in vars.conf.xml or using $${local_ip4}. The profile will use the IP it is listening on for Via Contact etc. Set the sip-ip and rtp-ip parameters of the relevant sofia profile (in sip_profiles/) to the 107.1.nnn.n address. Since the server has the IP assigned directly (not unusual) you do -not- need to set ext-sip-ip and ext-rtp-ip since you're not behind NAT. On 17 April 2015 at 15:54, T Fred Farmington <tfred31@yahoo.com> wrote:
My FreeSWITCH server has 2 IP addresses An Internal 192.168,nnn.nn & an External 107,1.nnn.nn This is accomplished via network 'magic' (I'm not the network guy) using a single network card
And, in general, it works. I can Receive Outside calls from our SIP Trunk vendor But I can Not Send calls Out through the SIP Trunk vendor.
The calls are being blocked (403 'Forbidden") by the vendor due to the contents of our SIP Header packet.
Within the Via: part of the packet it shows: Via: SIP/2.0/UDP 192.168.nnn.nn;rport;branch=z9hG4bKQKa7126jtX4mm They need it to show the External IP here.
I have even tried setting sip_via_host = 107.1.nnn.nn but that did not change what it being sent.
How/Where can I get this to change?
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Thank you for your reply. Currently the external_sip_ip and internal_rtp_ip are set to the default stun domain I went into the sip_profiles/external.xml Previous settings: sip_ip = $$(local_ip_v4) rtp_ip = $$(local_ip_v4) changed settings to Public IP sip_ip = 107.1.nnn.nnn rtp_ip = 107.1.nnn.nnn I then did a ReStart of the FS Service so as to ensure that the new values were applied With those new values in place I attempted a call to the outside which failed as before (fast-busy) I then went into the log and did not see where FS made any attempt to send any SIP Header to the SIP Trunk at all. That is a change since before it was sending, but just not the correct info. When I returned the values to their previous settings (ReStart, et al) , the log showed that the SIP Header was indeed sent, just with the wrong info. Just for FYI purposes: This FS Server is running Windows 2012 R2 Also in sofia status I see where the In-House IP is showing and not the Public IP (I don't know if this is correct or not)
I made some closely associated guesses and found the answer.... In the conf/sip_profiles/external.xml I needed to change the ext-sip-ip & ext-rtp-ip from their default of "auto-nat" to the Public IP Once that was done, the SIP Header Via contents showed correctly and the calls were going out. Thank you for your replies -------------------------------------------- On Fri, 4/17/15, Steven Ayre <steveayre@gmail.com> wrote: Subject: Re: [Freeswitch-users] SIP Header Via - IP Wrong To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Date: Friday, April 17, 2015, 11:16 AM A profile has to listen on a specific IP address. If you want to listen on both IPs you'll need a profile on each one. It's probably easiest in your setup to put the IP addresses into the profile directly rather than setting them in vars.conf.xml or using $${local_ip4}. The profile will use the IP it is listening on for Via Contact etc. Set the sip-ip and rtp-ip parameters of the relevant sofia profile (in sip_profiles/) to the 107.1.nnn.n address. Since the server has the IP assigned directly (not unusual) you do -not- need to set ext-sip-ip and ext-rtp-ip since you're not behind NAT. On 17 April 2015 at 15:54, T Fred Farmington <tfred31@yahoo.com> wrote: My FreeSWITCH server has 2 IP addresses An Internal 192.168,nnn.nn & an External 107,1.nnn.nn This is accomplished via network 'magic' (I'm not the network guy) using a single network card And, in general, it works. I can Receive Outside calls from our SIP Trunk vendor But I can Not Send calls Out through the SIP Trunk vendor. The calls are being blocked (403 'Forbidden") by the vendor due to the contents of our SIP Header packet. Within the Via: part of the packet it shows: Via: SIP/2.0/UDP 192.168.nnn.nn;rport;branch=z9hG4bKQKa7126jtX4mm They need it to show the External IP here. I have even tried setting sip_via_host = 107.1.nnn.nn but that did not change what it being sent. How/Where can I get this to change? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 (2)
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Steven Ayre
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T Fred Farmington