Inbound calls mis-routing routing to internal extension with external IP?
Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@*<externalip>* :46072 Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@<external_ip>:5080 RUNNING (0) external::<uuid> gateway sip:<flowrouteid>@sip.flowroute.com REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@<internal_ip>:5060 RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian
It is probably sending call to your extension@domainname, if your external ip is your domain name then you will see the external ip. Is your endpoint registered? How often is it set to reregister? Does your router have a sip alg? Are the ports opened in your firewall? Very hard to guess without a log of a call with debug enabled. sofia global sip trace on Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian Chow Sent: Wednesday, February 24, 2016 10:00 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP? Hello all! I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT. My Setup: Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance. SIP Provider: Flowroute NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively. I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine). When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip? I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@<externalip>:46072 Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ? Sofia Status says: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external profile sip:mod_sofia@<external_ip>:5080<sip:mod_sofia@%3cexternal_ip%3e:5080> RUNNING (0) external::<uuid> gateway sip:<flowrouteid>@sip.flowroute.com<http://sip.flowroute.com> REGED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@<internal_ip>:5060<sip:mod_sofia@%3cinternal_ip%3e:5060> RUNNING (0) ================================================================================================= If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be. Thanks, -Brian
Debug console output: https://pastebin.freeswitch.org/24572 My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.
Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600
Does your router have a sip alg? Are the ports opened in your firewall?
ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip.
Thanks! -Brian On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max@nysolutions.com> wrote:
It is probably sending call to your extension@domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
[image: cid:image001.jpg@01C72F94.9EE45D60] <http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Chow *Sent:* Wednesday, February 24, 2016 10:00 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?
Hello all!
I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT.
My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute
NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.
I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).
When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?
I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@ *<externalip>*:46072
Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ?
Sofia Status says:
Name Type Data State
=================================================================================================
external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0)
external profile sip:mod_sofia@<external_ip>:5080 RUNNING (0)
external::<uuid> gateway sip:<flowrouteid>@ sip.flowroute.com REGED
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
internal profile sip:mod_sofia@<internal_ip>:5060 RUNNING (0)
=================================================================================================
If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be.
Thanks,
-Brian
_________________________________________________________________________ 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
Its just the channel name so don't get tripped up by that, its just named with some default data about the session. On Wed, Feb 24, 2016 at 11:33 PM, Brian Chow <freeswitch@digitaldescent.net> wrote:
Debug console output: https://pastebin.freeswitch.org/24572
My fqdn only resolves inside the network, but the external-ip is configured in FS to use stun, which reports my external ip.
Is your endpoint registered? How often is it set to reregister?
Yes, and the zoiper default registration expiry is 3600
Does your router have a sip alg? Are the ports opened in your firewall?
ish, no GUI for it, but I did make sure to disable it from the command line (Ubiquiti Edgerouter). No firewall ports are explicitly open, but allow bi-directional with existing states. I do get 2 way audio when calling out, and I also can hear the system forwarding my call in to voicemail; the voicemail message and prompts. I know asterisk is different from FS, but my asterisk experience has proven that at least with asterisk, port forwarding isn't needed. Also, unfortunately, I cannot get a static ip.
Thanks!
-Brian
On Wed, Feb 24, 2016 at 8:31 PM, Moishe Grunstein <max@nysolutions.com> wrote:
It is probably sending call to your extension@domainname, if your external ip is your domain name then you will see the external ip.
Is your endpoint registered? How often is it set to reregister?
Does your router have a sip alg? Are the ports opened in your firewall?
Very hard to guess without a log of a call with debug enabled.
sofia global sip trace on
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
[image: cid:image001.jpg@01C72F94.9EE45D60] <http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Brian Chow *Sent:* Wednesday, February 24, 2016 10:00 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Inbound calls mis-routing routing to internal extension with external IP?
Hello all!
I'm new to freeswitch, so I'm sure this is just a newbie configuration error. Sorry if it's been answered a million times, my google searches always just bring up the standard NAT configuration pages. I've already followed the confluence page on configuring NAT.
My Setup:
Freeswitch 1.6 and FusionPBX installed on a virtualized debian 8.3 instance.
SIP Provider: Flowroute
NAT: extension and freeswitch are on the same network, both of which are behind NAT and connecting to flowroute. I configured external sip and rtp to use stun entries. External profile is using $${external_rtp/sip_ip} for ext-rtp/sip respectively.
I have one DID configured to go directly to my one extension. My extension can register just fine. My extension can dial out. When I call my cell, the call connects and I have 2 way audio. When I dial my DID from my cell, I can see the call hitting the FS server, but instead of ringing my extension, it goes straight to my extensions voicemail (which I can just fine).
When I look the at the console, it appears (sorry if this is wrong, I'm only a day into free switch) that FS is attempting to route the call to my extension...@ my external ip?
I see this line in the console: 2016-02-24 18:56:10.453983 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/<extension>@ *<externalip>*:46072
Shouldn't that read sofia/internal/<extentsion>@<ip_of_extension> ?
Sofia Status says:
Name Type Data State
=================================================================================================
external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0)
external profile sip:mod_sofia@<external_ip>:5080 RUNNING (0)
external::<uuid> gateway sip:<flowrouteid>@ sip.flowroute.com REGED
internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)
internal profile sip:mod_sofia@<internal_ip>:5060 RUNNING (0)
=================================================================================================
If I'm completely off base here, can anyone recommend where I can start looking to change/troubleshoot the issue? I feel like it's just me missing something, I just can't determine what that might be.
Thanks,
-Brian
_________________________________________________________________________ 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
_________________________________________________________________________ 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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
participants (3)
-
Brian Chow
-
Brian West
-
Moishe Grunstein