Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]
What deps were missing? On Sunday, January 4, 2015, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <javascript:_e(%7B%7D,'cvml','Richard.Adams@STENTOFON.COM.AU');> ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Hello; It started with Git. Then autoconf. That's when I realized I should just proceed manually. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Brian West [mailto:brian@freeswitch.org] Sent: Monday, January 05, 2015 12:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required What deps were missing? On Sunday, January 4, 2015, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<javascript:_e(%7B%7D,'cvml','Richard.Adams@STENTOFON.COM.AU');> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] -- Brian West brian@freeswitch.org<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
Huh? On Sunday, January 4, 2015, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
Hello;
It started with Git. Then autoconf. That's when I realized I should just proceed manually.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <javascript:_e(%7B%7D,'cvml','Richard.Adams@STENTOFON.COM.AU');> ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Brian West [mailto:brian@freeswitch.org <javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');>] *Sent:* Monday, January 05, 2015 12:31 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
What deps were missing?
On Sunday, January 4, 2015, Richard Adams <Richard.Adams@stentofon.com.au <javascript:_e(%7B%7D,'cvml','Richard.Adams@stentofon.com.au');>> wrote:
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
--
*Brian West* brian@freeswitch.org <javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');>
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
This is really strange, I've been installing FS for years, not a single issue... please use: http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide Note the dependencies: sudo apt-get install libcurl4-openssl-dev libexpat1-dev libssl-dev libtiff4-dev libx11-dev unixodbc-dev \ python2.6-dev zlib1g-dev libzrtpcpp-dev libasound2-dev libogg-dev libvorbis-dev libperl-dev \ libgdbm-dev libdb-dev python-dev uuid-dev bison autoconf g++ libncurses-dev follow the instructions there, you should be fine. David On Mon, Jan 5, 2015 at 2:56 AM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
Hello;
It started with Git. Then autoconf. That's when I realized I should just proceed manually.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Brian West [mailto:brian@freeswitch.org] *Sent:* Monday, January 05, 2015 12:31 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
What deps were missing?
On Sunday, January 4, 2015, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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
-- DVG -- Imagination is more important than knowledge Albert Einstein
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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 Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]
Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]
When you compile from source its up to you yo install any prerequisite packages for the distro You are using There are some make files and instructions on confluence to help you with this process. Or you can use the precompiled debian packages. Refer to confluence for setting up the sources.list Sent from my iPhone
On Jan 4, 2015, at 7:57 PM, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source.
Regards,
Richard Adams Technical Manager
STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ● WWW.DNH.NO <image001.jpg>
From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified <image002.jpg> 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 Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams Technical Manager
STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU WWW.ZENITEL.COM ● WWW.DNH.NO <image001.jpg>
_________________________________________________________________________ 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
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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 Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850]
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP's are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1)Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT <https://freeswitch.org/confluence/display/FREESWITCH/NAT> and many other NAT pages in the wiki
2)Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
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UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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
Have you done a SIP trace and compared the RTP packets between a normal call and the diverted call? Also look for packets coming from the outside to the wrong ports, or not showing up at all. On 5/01/2015 3:25 PM, Richard Adams wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:01 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP's are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1)Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT <https://freeswitch.org/confluence/display/FREESWITCH/NAT> and many other NAT pages in the wiki
2)Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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The Invite packet is identical in either the successful call and the unsuccessful call. My provider mentions the following; Hi Richard, I can't really say based on the SIP logs given I've looked at a couple samples & they look similar, but investigating our media logs shows me why. For the successful call the initial media traffic is sent to 192.168.34.2:21583, after around four seconds we receive a request to change the IP/port to 123.243.120.79:21582 which is correct. Whereas on the failed sample we only receive a request to send traffic through to 192.168.34.2:16584, the port is correct but the IP isn't. Hence why the media is never received. The two samples above are for the one you provided & my initial sample. The overall issue is the incorrect IP being sent in the initial Invite or 200 OK, so even though regular inbound/outbound calls may be working it is recommended to fix the SDP attributes as it will still cause issues. The SDP is; v=0 o=FreeSWITCH 1416781294 1416781295 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 21582 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Can I force the SDP changes? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au] Sent: Monday, January 05, 2015 4:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Have you done a SIP trace and compared the RTP packets between a normal call and the diverted call? Also look for packets coming from the outside to the wrong ports, or not showing up at all. On 5/01/2015 3:25 PM, Richard Adams wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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
Richard, What router are you using ?? Does it have a SIP ALG ( or SIP Helper or similar ?? ) if so turn it off and try again. Jay On 5 January 2015 at 15:50, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
The Invite packet is identical in either the successful call and the unsuccessful call.
My provider mentions the following;
Hi Richard,
I can't really say based on the SIP logs given I've looked at a couple samples & they look similar, but investigating our media logs shows me why.
For the successful call the initial media traffic is sent to 192.168.34.2:21583, after around four seconds we receive a request to change the IP/port to 123.243.120.79:21582 which is correct.
Whereas on the failed sample we only receive a request to send traffic through to 192.168.34.2:16584, the port is correct but the IP isn't. Hence why the media is never received.
The two samples above are for the one you provided & my initial sample. The overall issue is the incorrect IP being sent in the initial Invite or 200 OK, so even though regular inbound/outbound calls may be working it is recommended to fix the SDP attributes as it will still cause issues.
The SDP is;
v=0
o=FreeSWITCH 1416781294 1416781295 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 21582 RTP/AVP 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Can I force the SDP changes?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:41 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Have you done a SIP trace and compared the RTP packets between a normal call and the diverted call? Also look for packets coming from the outside to the wrong ports, or not showing up at all.
On 5/01/2015 3:25 PM, Richard Adams wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 ( sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [ sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [ sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 ( sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 ( sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 ( sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 ( sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au <sms@icefire.qza.net.au>] *Sent:* Monday, January 05, 2015 4:01 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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
_________________________________________________________________________ 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
-- Sincerely Jay
Netcomm NB16WV SIP NAT is off. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 5:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard, What router are you using ?? Does it have a SIP ALG ( or SIP Helper or similar ?? ) if so turn it off and try again. Jay On 5 January 2015 at 15:50, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: The Invite packet is identical in either the successful call and the unsuccessful call. My provider mentions the following; Hi Richard, I can't really say based on the SIP logs given I've looked at a couple samples & they look similar, but investigating our media logs shows me why. For the successful call the initial media traffic is sent to 192.168.34.2:21583<http://192.168.34.2:21583>, after around four seconds we receive a request to change the IP/port to 123.243.120.79:21582<http://123.243.120.79:21582> which is correct. Whereas on the failed sample we only receive a request to send traffic through to 192.168.34.2:16584<http://192.168.34.2:16584>, the port is correct but the IP isn't. Hence why the media is never received. The two samples above are for the one you provided & my initial sample. The overall issue is the incorrect IP being sent in the initial Invite or 200 OK, so even though regular inbound/outbound calls may be working it is recommended to fix the SDP attributes as it will still cause issues. The SDP is; v=0 o=FreeSWITCH 1416781294 1416781295 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 21582 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Can I force the SDP changes? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>] Sent: Monday, January 05, 2015 4:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Have you done a SIP trace and compared the RTP packets between a normal call and the diverted call? Also look for packets coming from the outside to the wrong ports, or not showing up at all. On 5/01/2015 3:25 PM, Richard Adams wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<mailto:sofia/internal/sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:sofia/external/2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:sofia/internal/127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:sofia/external/61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:sofia/external/8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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 -- Sincerely Jay
Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/ 127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/ 61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:01 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Sincerely Jay
I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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 -- Sincerely Jay
That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/ 127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/ 61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:01 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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
_________________________________________________________________________ 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
--
Sincerely
Jay
_________________________________________________________________________ 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
-- Sincerely Jay
.2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798<tel:0404058798> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798<tel:0404058798>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798<tel:0404058798> [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay
it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/ 127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/ 61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:01 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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
_________________________________________________________________________ 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
--
Sincerely
Jay
_________________________________________________________________________ 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
--
Sincerely
Jay
_________________________________________________________________________ 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
-- Sincerely Jay
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ To: <sip:0404058798@sip20.mynetfone.com.au> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600@sip20.mynetfone.com.au>;tag=egN14DrUFcD6j To: <sip:0404058798@sip20.mynetfone.com.au> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798<tel:0404058798> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798<tel:0404058798>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798<tel:0404058798> [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay
Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/ 127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/ 61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Francis [mailto:sms@icefire.qza.net.au] *Sent:* Monday, January 05, 2015 4:01 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com <max@nysolutions.com>]
*Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2) Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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--
Sincerely
Jay
_________________________________________________________________________ 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
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--
Sincerely
Jay
_________________________________________________________________________ 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
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--
Sincerely
Jay
_________________________________________________________________________ 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
-- Sincerely Jay
I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] has been answered 2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>] has been answered 2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered 2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>] has been answered 2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221] 2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79>) Ended 2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79<mailto:2057@123.243.120.79> [CS_DESTROY] 2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY] 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>) Ended 2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [CS_DESTROY] 2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798<tel:0404058798> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798<tel:0404058798>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798<tel:0404058798> [CS_DESTROY] 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83>) Ended 2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83<mailto:61397296600@125.213.160.83> [CS_DESTROY] 2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221] 2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> has executed the last dialplan instruction, hanging up. 2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING] 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79>) Ended 2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79<mailto:8363@123.243.120.79> [CS_DESTROY] Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Francis [mailto:sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>] Sent: Monday, January 05, 2015 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker. If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases. *However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne. On 5/01/2015 12:35 PM, Richard Adams wrote: Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection. I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully. We're using UPNP, and the ports are forwarded correctly. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 1:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Hello, 1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call. 2. Compile from source. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Monday, January 05, 2015 12:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required 1) Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki 2) Which install method did you use? did you compile from source? Did you create a init.d? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Sunday, January 4, 2015 8:19 PM To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Help required Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC. 3 issues; 1. No audio on calls when an external call is bridged to another external location. We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call. 2. No autostart Freeswitch on system boot for Debian 7 and current build. To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up. 3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually. If I can get some guidance please, and I'm happy to provide whatever logging is required. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 -- Sincerely Jay
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au <mailto:Richard.Adams@stentofon.com.au>> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798 <tel:0404058798>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au> SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600 <tel:0397296600>" <sip:0397296600 <tel:0397296600>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ
To: <sip:0404058798 <tel:0404058798>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600 <tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83 <mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600 <tel:0397296600>" <sip:0397296600 <tel:0397296600>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798 <tel:0404058798>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au> SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600 <tel:0397296600>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j
To: <sip:0404058798 <tel:0404058798>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600 <tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600 <tel:0397296600>@sip20.mynetfone.com.au <http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 ? FAX: +61 3 9729 0099 ? MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*jay binks [mailto:jaybinks@gmail.com <mailto:jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au <mailto:Richard.Adams@stentofon.com.au>> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*jay binks [mailto:jaybinks@gmail.com <mailto:jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au <mailto:Richard.Adams@stentofon.com.au>> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*jay binks [mailto:jaybinks@gmail.com <mailto:jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au <mailto:Richard.Adams@stentofon.com.au>> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600 <tel:61397296600>@125.213.160.83 <http://125.213.160.83> [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <tel:61397296600> <61397296600 <tel:61397296600>>->0386447200 <tel:0386447200> in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600 <tel:61397296600>@125.213.160.83 <http://125.213.160.83> to XML[0386447200 <tel:%5B0386447200>@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <tel:61397296600> <61397296600 <tel:61397296600>>->0386447200 <tel:0386447200> in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 <http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100 <http://sip:122@192.168.34.20:2100>!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600 <tel:61397296600>@125.213.160.83 <http://125.213.160.83>!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600 <tel:61397296600>@125.213.160.83 <http://125.213.160.83>!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2>!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2>!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 <http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100 <http://sip:122@192.168.34.20:2100>) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 <http://sip:122@192.168.34.20:2100> [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 <tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798 <tel:0404058798>!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798 <tel:0404058798>] has been answered
2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83 <mailto:61397296600@125.213.160.83>] has been answered
2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered
2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2>] has been answered
2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 <mailto:2057@123.243.120.79> [30909a08-949a-11e4-9736-192279674221]
2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 <mailto:2057@123.243.120.79> has executed the last dialplan instruction, hanging up.
2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 <mailto:2057@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79 <mailto:2057@123.243.120.79>) Ended
2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 <mailto:2057@123.243.120.79> [CS_DESTROY]
2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2> [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2>) Ended
2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 <mailto:127@192.168.34.2> [CS_DESTROY]
2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83 <mailto:61397296600@125.213.160.83> [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 <tel:0404058798> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798 <tel:0404058798>) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 <tel:0404058798> [CS_DESTROY]
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83 <mailto:61397296600@125.213.160.83>) Ended
2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 <mailto:61397296600@125.213.160.83> [CS_DESTROY]
2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 <mailto:8363@123.243.120.79> [39838eb8-949a-11e4-9739-192279674221]
2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 <mailto:8363@123.243.120.79> has executed the last dialplan instruction, hanging up.
2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 <mailto:8363@123.243.120.79> [CS_EXECUTE] [NORMAL_CLEARING]
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79 <mailto:8363@123.243.120.79>) Ended
2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 <mailto:8363@123.243.120.79> [CS_DESTROY]
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Francis [mailto:sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>] *Sent:* Monday, January 05, 2015 4:01 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.
If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.
*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.
On 5/01/2015 12:35 PM, Richard Adams wrote:
Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.
I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.
We're using UPNP, and the ports are forwarded correctly.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 1:23 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Well if internal IP's are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US <http://888.IPPBX.US> *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:58 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Hello,
1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.
2. Compile from source.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
*From:*Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Monday, January 05, 2015 12:34 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
1)Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki
2)Which install method did you use? did you compile from source? Did you create a init.d?
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US <http://888.IPPBX.US> *Service Request Email: support@nysolutions.com <mailto:support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Richard Adams *Sent:* Sunday, January 4, 2015 8:19 PM *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* [Freeswitch-users] Help required
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
/Technical Manager/
*STENTOFON AUSTRALIA*? UNIT 2 / 670 MOUNTAIN HIGHWAY ? BAYSWATER VIC 3153 ? AUSTRALIA
PHONE: +61 3 9729 6600 <tel:%2B61%203%209729%206600> ? FAX: +61 3 9729 0099 <tel:%2B61%203%209729%200099> ? MOBILE: +61 404 058 798 <tel:%2B61%20404%20058%20798>
Richard.Adams@STENTOFON.COM.AU <mailto:Richard.Adams@STENTOFON.COM.AU>? WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/>?WWW.DNH.NO <http://www.dnh.no/>
cid:image001.jpg@01CD7927.5758B850
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org <mailto:consulting@freeswitch.org>
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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
--
Sincerely
Jay
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
Sincerely
Jay
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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
--
Sincerely
Jay
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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
--
Sincerely
Jay
_________________________________________________________________________ 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
Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au> написал:
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them.
On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com <jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 <%2B61%203%209729%206600> ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798]
_________________________________________________________________________ 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
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I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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 ...
Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same.
The SDP still uses the private IP of the PBX.
What do I need to set to re-write the O and S records in the SDP to my public IP?
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Daniel Ivanov [mailto:sertys@gmail.com] *Sent:* Monday, January 05, 2015 7:59 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Just set the NDLB_force_rport param on your mynetfone profile and be done with it.
5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au> написал:
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them.
On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com <jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798]
_________________________________________________________________________ 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
SHHH don't tell them to look at the box for the credentials, Its a test to see if they are paying attention. :) On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with.
you can use the freeswitch pastebin: http://pastebin.freeswitch.org
Instructions/credentials are included on the box that pops up.
At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router.
On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same.
The SDP still uses the private IP of the PBX.
What do I need to set to re-write the O and S records in the SDP to my public IP?
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Daniel Ivanov [mailto:sertys@gmail.com] *Sent:* Monday, January 05, 2015 7:59 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Just set the NDLB_force_rport param on your mynetfone profile and be done with it.
5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au> написал:
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them.
On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com <jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798]
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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I'm stupid, but not quite that stupid... Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Brian West [mailto:brian@freeswitch.org] Sent: Tuesday, January 06, 2015 11:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required SHHH don't tell them to look at the box for the credentials, Its a test to see if they are paying attention. :) On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge <blasterjr@gmail.com<mailto:blasterjr@gmail.com>> wrote: Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto:brian@freeswitch.org> [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
LOL, Wasn't implying you were... I actually do use that as a test to see if someone is paying attention or trying to go too fast without fully understanding whats going on.... missing that usually means something else was missed along the way. On Mon, Jan 5, 2015 at 6:28 PM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
I'm stupid, but not quite that stupid...
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Brian West [mailto:brian@freeswitch.org] *Sent:* Tuesday, January 06, 2015 11:05 AM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
SHHH don't tell them to look at the box for the credentials, Its a test to see if they are paying attention. :)
On Mon, Jan 5, 2015 at 5:59 PM, Chris Tunbridge <blasterjr@gmail.com> wrote:
Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with.
you can use the freeswitch pastebin: http://pastebin.freeswitch.org
Instructions/credentials are included on the box that pops up.
At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router.
On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same.
The SDP still uses the private IP of the PBX.
What do I need to set to re-write the O and S records in the SDP to my public IP?
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Daniel Ivanov [mailto:sertys@gmail.com] *Sent:* Monday, January 05, 2015 7:59 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Just set the NDLB_force_rport param on your mynetfone profile and be done with it.
5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au> написал:
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them.
On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com <jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798]
_________________________________________________________________________ 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
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_________________________________________________________________________ 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
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_________________________________________________________________________ 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
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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 *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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
<param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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 Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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 enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required <param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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
You should on your internal profile. It actually is in there just uncomment it. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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 Richard Adams Sent: Monday, January 5, 2015 8:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required <param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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
Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" <Richard.Adams@stentofon.com.au> escreveu:
I enabled this on the External.xml to no change.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Moishe Grunstein [mailto:max@nysolutions.com] *Sent:* Tuesday, January 06, 2015 11:31 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
<param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway.
Thanks,
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US *Service Request Email: support@nysolutions.com <support@nysolutions.com> *
Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 <freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Richard Adams *Sent:* Monday, January 5, 2015 7:15 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
https://pastebin.freeswitch.org/23822 for Internal.xml
https://pastebin.freeswitch.org/23823 for External.xml
https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Chris Tunbridge [mailto:blasterjr@gmail.com <blasterjr@gmail.com>]
*Sent:* Tuesday, January 06, 2015 11:00 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with.
you can use the freeswitch pastebin: http://pastebin.freeswitch.org
Instructions/credentials are included on the box that pops up.
At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router.
On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams < Richard.Adams@stentofon.com.au> wrote:
I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same.
The SDP still uses the private IP of the PBX.
What do I need to set to re-write the O and S records in the SDP to my public IP?
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* Daniel Ivanov [mailto:sertys@gmail.com] *Sent:* Monday, January 05, 2015 7:59 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Just set the NDLB_force_rport param on your mynetfone profile and be done with it.
5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au> написал:
There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them.
On 5/01/2015 6:11 PM, Richard Adams wrote:
I said practically. If they're both broken, then they're practically the same :)
I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.
Where else do I need to set the public IP to get the SDP to report correctly?
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com <jaybinks@gmail.com>] *Sent:* Monday, January 05, 2015 6:09 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Not identical.
the FROM's differ one has X-Serialnumber & P-Key-Flags headers.
Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems.
that NEEDS To have your external, REAL IP Address.
Do you know what that IP does in the SDP ??
Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )
Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P )
On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.
The packets are practically identical.
send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr
Max-Forwards: 33
From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;tag=D7U82j7Qj3pKQ
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: b36af627-0f48-1233-04b8-00188b436d1c
CSeq: 69883976 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
Diversion: <sip:86447200@125.213.160.83>;reason=unconditional
X-FS-Support: update_display,send_info
Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17482 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:
------------------------------------------------------------------------
INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p
Max-Forwards: 69
From: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;tag=egN14DrUFcD6j
To: <sip:0404058798@sip20.mynetfone.com.au>
Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c
CSeq: 69884009 INVITE
Contact: <sip:0397296600@123.243.120.79:5080 ;transport=udp;gw=MyNetFone>
User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 270
X-Serialnumber: 0004133887C7
P-Key-Flags: keys="3"
X-FS-Support: update_display,send_info
Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au
;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2
s=FreeSWITCH
c=IN IP4 192.168.34.2
t=0 0
m=audio 17766 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:35 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
it might just be easier to pastebin your config files , specifically the SIP profiles for a start.
On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
.2 is the PBX.
Bypass media is not on.
I'll look at the RTP-IP.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 5:09 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
That is kinda what happens when you post IP Address in a mailing list or online :(
You can rest assured MyNetFone will also be getting them to their IP address.
OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.
What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2
if .34.2 is a phone handset then id guess its closer to option #3
Jay
On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
I have attached a packet trace from a failed call.
We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.
We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.
Some nefarious people are on this mailing list, as I'm now under attack. Hooray!
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
*From:* jay binks [mailto:jaybinks@gmail.com] *Sent:* Monday, January 05, 2015 4:48 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Help required
Im coming in late to the party here.
Do you have any packet captures of this ???
It sounds like your trying to bypass the media and get out of the RTP stream.
Nobody is going to do that for you :)
Also, from your logs I can see your on a TPG service.
A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.
I hope that is just the service your using for your testing ??
Jay Binks
On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:
UPNP is now off. Same result.
The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]
2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default
2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public
2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]
2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default
2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]
2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/ sip:122@192.168.34.20:2100!
2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!
2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!
2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!
2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]
2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]
2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended
2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]
2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]
2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!
2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/ 0404058798]
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sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON@192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON@192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600@sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio@tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com<mailto:max@nysolutions.com>] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required <param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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
Are you on IRC? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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 Richard Adams Sent: Monday, January 5, 2015 8:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON@192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON@192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600@sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio@tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com<mailto:max@nysolutions.com>] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required <param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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
Yes, am now as STENTOFON Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com] Sent: Tuesday, January 06, 2015 1:00 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Are you on IRC? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 8:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@xxx.243.120.79:5080 RUNNING (0) external::dialalphacomcompletenetwork gateway sip:STENTOFON@192.168.34.5:5080 NOREG external::dialalphacom gateway sip:STENTOFON@192.168.34.4:5080 NOREG external::MyNetFone gateway sip:0397296600@sip20.mynetfone.com.au NOREG 192.168.34.2 alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@xxx.243.120.79:5060 RUNNING (0) ================================================================================================= 3 profiles 1 alias Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798 Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Wesley Akio [mailto:wesleyakio@tuntscorp.com] Sent: Tuesday, January 06, 2015 12:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Could you please post the result to sofia status? Best, Wesley Akio Sent from mobile Em 05/01/2015 23:14, "Richard Adams" <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> escreveu: I enabled this on the External.xml to no change. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Moishe Grunstein [mailto:max@nysolutions.com<mailto:max@nysolutions.com>] Sent: Tuesday, January 06, 2015 11:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required <param name="NDLB-force-rport" value="true"/> belongs on the profile not sip gateway. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650<tel:212.400.7650> 888.IPPBX.US<http://888.IPPBX.US> Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [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> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams Sent: Monday, January 5, 2015 7:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required https://pastebin.freeswitch.org/23822 for Internal.xml https://pastebin.freeswitch.org/23823 for External.xml https://pastebin.freeswitch.org/23824 for 00_dialmynetfone.xml Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Chris Tunbridge [mailto:blasterjr@gmail.com] Sent: Tuesday, January 06, 2015 11:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Richard someone else asked this, and i think they were correct in asking, can you please pastebin your config files, specifically the ones in conf/sip_profiles/ as that'll give us the best information to help you with. you can use the freeswitch pastebin: http://pastebin.freeswitch.org Instructions/credentials are included on the box that pops up. At this point to me, it sounds like improper (or missing) configuration, OR its getting messed with by the router. On Mon, Jan 5, 2015 at 3:16 PM, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I tried setting NDLB_force_rport on my mynetfone profile, but the result is the same. The SDP still uses the private IP of the PBX. What do I need to set to re-write the O and S records in the SDP to my public IP? o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: Daniel Ivanov [mailto:sertys@gmail.com<mailto:sertys@gmail.com>] Sent: Monday, January 05, 2015 7:59 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Just set the NDLB_force_rport param on your mynetfone profile and be done with it. 5 янв. 2015 г. 10:30 пользователь "Francis" <sms@icefire.qza.net.au<mailto:sms@icefire.qza.net.au>> написал: There are some exotic NAT settings on the wiki that will _force_ certain behaviours, which may get things working and give you a starting point to work back from. Bear in mind that these settings aren't really fixes, just ways of telling FS to treat devices as stoopid and do the thinking for them. On 5/01/2015 6:11 PM, Richard Adams wrote: I said practically. If they're both broken, then they're practically the same :) I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile. Where else do I need to set the public IP to get the SDP to report correctly? Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com] Sent: Monday, January 05, 2015 6:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Not identical. the FROM's differ one has X-Serialnumber & P-Key-Flags headers. Both of them have bad IP's in the SDP ( 192.168.34.2 ) this will be key here.. fix this and it may fix all your problems. that NEEDS To have your external, REAL IP Address. Do you know what that IP does in the SDP ?? Thats what tells My Net Fone where to send the audio. how can they route to that IP address, its not globally routable. so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing ) Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79 ( XXX Added to protect you from more attacks :P ) On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile. The packets are practically identical. send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr Max-Forwards: 33 From: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=D7U82j7Qj3pKQ To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: b36af627-0f48-1233-04b8-00188b436d1c CSeq: 69883976 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 Diversion: <sip:86447200@125.213.160.83<mailto:sip%3A86447200@125.213.160.83>>;reason=unconditional X-FS-Support: update_display,send_info Remote-Party-ID: "0397296600<tel:0397296600>" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17482 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905: ------------------------------------------------------------------------ INVITE sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au> SIP/2.0 Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p Max-Forwards: 69 From: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;tag=egN14DrUFcD6j To: <sip:0404058798<tel:0404058798>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>> Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c CSeq: 69884009 INVITE Contact: <sip:0397296600<tel:0397296600>@123.243.120.79:5080;transport=udp;gw=MyNetFone> User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 0004133887C7 P-Key-Flags: keys="3" X-FS-Support: update_display,send_info Remote-Party-ID: "Richard" <sip:0397296600<tel:0397296600>@sip20.mynetfone.com.au<http://sip20.mynetfone.com.au>>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2 s=FreeSWITCH c=IN IP4 192.168.34.2 t=0 0 m=audio 17766 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required it might just be easier to pastebin your config files , specifically the SIP profiles for a start. On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: .2 is the PBX. Bypass media is not on. I'll look at the RTP-IP. Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 5:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required That is kinda what happens when you post IP Address in a mailing list or online :( You can rest assured MyNetFone will also be getting them to their IP address. OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something. What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2 if .34.2 is a phone handset then id guess its closer to option #3 Jay On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: I have attached a packet trace from a failed call. We use TPG. The service is exceptional, and we've had only one downtime event in 5 years. We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line. Some nefarious people are on this mailing list, as I'm now under attack. Hooray! Regards, Richard Adams Technical Manager STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA PHONE: +61 3 9729 6600<tel:%2B61%203%209729%206600> ● FAX: +61 3 9729 0099<tel:%2B61%203%209729%200099> ● MOBILE: +61 404 058 798<tel:%2B61%20404%20058%20798> Richard.Adams@STENTOFON.COM.AU<mailto:Richard.Adams@STENTOFON.COM.AU> ● WWW.STENTOFON.COM.AU<http://www.stentofon.com.au/> WWW.ZENITEL.COM<http://www.zenitel.com/> ● WWW.DNH.NO<http://www.dnh.no/> [cid:image001.jpg@01CD7927.5758B850] From: jay binks [mailto:jaybinks@gmail.com<mailto:jaybinks@gmail.com>] Sent: Monday, January 05, 2015 4:48 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help required Im coming in late to the party here. Do you have any packet captures of this ??? It sounds like your trying to bypass the media and get out of the RTP stream. Nobody is going to do that for you :) Also, from your logs I can see your on a TPG service. A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev. I hope that is just the service your using for your testing ?? Jay Binks On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au<mailto:Richard.Adams@stentofon.com.au>> wrote: UPNP is now off. Same result. The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller. 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2> [253bea7c-949a-11e4-9703-192279674221] 2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default 2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221] 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> [25584398-949a-11e4-9710-192279674221] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context public 2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83> to XML[0386447200<tel:%5B0386447200>@default] 2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600<tel:61397296600> <61397296600<tel:61397296600>>->0386447200<tel:0386447200> in context default 2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [25594e96-949a-11e4-9727-192279674221] 2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>! 2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600<tel:61397296600>@125.213.160.83<http://125.213.160.83>! 2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200! 2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2<mailto:127@192.168.34.2>! 2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_CONSUME_MEDIA] [NO_ANSWER] 2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER] 2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100>) Ended 2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100<http://sip:122@192.168.34.20:2100> [CS_DESTROY] 2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798<tel:0404058798> [2b4cddb8-949a-11e4-9730-192279674221] 2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798<tel:0404058798>! 2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798<tel:0404058798>] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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
Please kindly describe your architecture is your server behind a nat? are your phones behind a nat? what codec are you using Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH 2015-01-04 20:19 GMT-05:00 Richard Adams <Richard.Adams@stentofon.com.au>:
Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.
3 issues;
1. No audio on calls when an external call is bridged to another external location.
We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.
2. No autostart Freeswitch on system boot for Debian 7 and current build.
To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.
3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.
If I can get some guidance please, and I'm happy to provide whatever logging is required.
Regards,
Richard Adams
*Technical Manager*
*STENTOFON AUSTRALIA* ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA
PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798
Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU <http://www.stentofon.com.au/>
WWW.ZENITEL.COM <http://www.zenitel.com/> ● WWW.DNH.NO <http://www.dnh.no/>
[image: cid:image001.jpg@01CD7927.5758B850]
_________________________________________________________________________ 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 (11)
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Brian West
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Chris Tunbridge
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Daniel Ivanov
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David Villasmil Govea
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Francis
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jay binks
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Ken Rice
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Luis Daniel Lucio Quiroz
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Moishe Grunstein
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Richard Adams
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Wesley Akio