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
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
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
● WWW.STENTOFON.COM.AU
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
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